libao2/ao_pcm.c

   1 /*
   2  * PCM audio output driver
   3  *
   4  * This file is part of MPlayer.
   5  *
   6  * MPlayer is free software; you can redistribute it and/or modify
   7  * it under the terms of the GNU General Public License as published by
   8  * the Free Software Foundation; either version 2 of the License, or
   9  * (at your option) any later version.
  10  *
  11  * MPlayer is distributed in the hope that it will be useful,
  12  * but WITHOUT ANY WARRANTY; without even the implied warranty of
  13  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
  14  * GNU General Public License for more details.
  15  *
  16  * You should have received a copy of the GNU General Public License along
  17  * with MPlayer; if not, write to the Free Software Foundation, Inc.,
  18  * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
  19  */
  20
  21 #include "config.h"
  22
  23 #include <stdio.h>
  24 #include <stdlib.h>
  25 #include <string.h>
  26
  27 #include "libavutil/common.h"
  28 #include "mpbswap.h"
  29 #include "subopt-helper.h"
  30 #include "libaf/af_format.h"
  31 #include "libaf/reorder_ch.h"
  32 #include "audio_out.h"
  33 #include "audio_out_internal.h"
  34 #include "mp_msg.h"
  35 #include "help_mp.h"
  36
  37
  38 static const ao_info_t info =
  39 {
  40         "RAW PCM/WAVE file writer audio output",
  41         "pcm",
  42         "Atmosfear",
  43         ""
  44 };
  45
  46 LIBAO_EXTERN(pcm)
  47
  48 extern int vo_pts;
  49
  50 static char *ao_outputfilename = NULL;
  51 static int ao_pcm_waveheader = 1;
  52 static int fast = 0;
  53
  54 #define WAV_ID_RIFF 0x46464952 /* "RIFF" */
  55 #define WAV_ID_WAVE 0x45564157 /* "WAVE" */
  56 #define WAV_ID_FMT  0x20746d66 /* "fmt " */
  57 #define WAV_ID_DATA 0x61746164 /* "data" */
  58 #define WAV_ID_PCM  0x0001
  59 #define WAV_ID_FLOAT_PCM  0x0003
  60
  61 struct WaveHeader
  62 {
  63         uint32_t riff;
  64         uint32_t file_length;
  65         uint32_t wave;
  66         uint32_t fmt;
  67         uint32_t fmt_length;
  68         uint16_t fmt_tag;
  69         uint16_t channels;
  70         uint32_t sample_rate;
  71         uint32_t bytes_per_second;
  72         uint16_t block_align;
  73         uint16_t bits;
  74         uint32_t data;
  75         uint32_t data_length;
  76 };
  77
  78 /* init with default values */
  79 static struct WaveHeader wavhdr;
  80 uint64_t data_length;
  81
  82 static FILE *fp = NULL;
  83
  84 // to set/get/query special features/parameters
  85 static int control(int cmd,void *arg){
  86     return -1;
  87 }
  88
  89 // open & setup audio device
  90 // return: 1=success 0=fail
  91 static int init(int rate,int channels,int format,int flags){
  92         int bits;
  93         opt_t subopts[] = {
  94           {"waveheader", OPT_ARG_BOOL, &ao_pcm_waveheader, NULL},
  95           {"file",       OPT_ARG_MSTRZ, &ao_outputfilename, NULL},
  96           {"fast",       OPT_ARG_BOOL, &fast, NULL},
  97           {NULL}
  98         };
  99         // set defaults
 100         ao_pcm_waveheader = 1;
 101
 102         if (subopt_parse(ao_subdevice, subopts) != 0) {
 103           return 0;
 104         }
 105         if (!ao_outputfilename){
 106           ao_outputfilename =
 107             strdup(ao_pcm_waveheader?"audiodump.wav":"audiodump.pcm");
 108         }
 109
 110         bits=8;
 111         switch(format){
 112         case AF_FORMAT_S32_BE:
 113             format=AF_FORMAT_S32_LE;
 114         case AF_FORMAT_S32_LE:
 115             bits=32;
 116             break;
 117         case AF_FORMAT_FLOAT_BE:
 118             format=AF_FORMAT_FLOAT_LE;
 119         case AF_FORMAT_FLOAT_LE:
 120             bits=32;
 121             break;
 122         case AF_FORMAT_S8:
 123             format=AF_FORMAT_U8;
 124         case AF_FORMAT_U8:
 125             break;
 126         case AF_FORMAT_AC3:
 127             bits=16;
 128             break;
 129         default:
 130             format=AF_FORMAT_S16_LE;
 131             bits=16;
 132             break;
 133         }
 134
 135         ao_data.outburst = 65536;
 136         ao_data.buffersize= 2*65536;
 137         ao_data.channels=channels;
 138         ao_data.samplerate=rate;
 139         ao_data.format=format;
 140         ao_data.bps=channels*rate*(bits/8);
 141
 142         wavhdr.riff = le2me_32(WAV_ID_RIFF);
 143         wavhdr.wave = le2me_32(WAV_ID_WAVE);
 144         wavhdr.fmt = le2me_32(WAV_ID_FMT);
 145         wavhdr.fmt_length = le2me_32(16);
 146         wavhdr.fmt_tag = le2me_16(format == AF_FORMAT_FLOAT_LE ? WAV_ID_FLOAT_PCM : WAV_ID_PCM);
 147         wavhdr.channels = le2me_16(ao_data.channels);
 148         wavhdr.sample_rate = le2me_32(ao_data.samplerate);
 149         wavhdr.bytes_per_second = le2me_32(ao_data.bps);
 150         wavhdr.bits = le2me_16(bits);
 151         wavhdr.block_align = le2me_16(ao_data.channels * (bits / 8));
 152
 153         wavhdr.data = le2me_32(WAV_ID_DATA);
 154         wavhdr.data_length=le2me_32(0x7ffff000);
 155         wavhdr.file_length = wavhdr.data_length + sizeof(wavhdr) - 8;
 156
 157         mp_msg(MSGT_AO, MSGL_INFO, MSGTR_AO_PCM_FileInfo, ao_outputfilename,
 158                (ao_pcm_waveheader?"WAVE":"RAW PCM"), rate,
 159                (channels > 1) ? "Stereo" : "Mono", af_fmt2str_short(format));
 160         mp_msg(MSGT_AO, MSGL_INFO, MSGTR_AO_PCM_HintInfo);
 161
 162         fp = fopen(ao_outputfilename, "wb");
 163         if(fp) {
 164                 if(ao_pcm_waveheader){ /* Reserve space for wave header */
 165                         fwrite(&wavhdr,sizeof(wavhdr),1,fp);
 166                 }
 167                 return 1;
 168         }
 169         mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_PCM_CantOpenOutputFile,
 170                ao_outputfilename);
 171         return 0;
 172 }
 173
 174 // close audio device
 175 static void uninit(int immed){
 176
 177         if(ao_pcm_waveheader){ /* Rewrite wave header */
 178                 if (fseek(fp, 0, SEEK_SET) != 0)
 179                         mp_msg(MSGT_AO, MSGL_ERR, "Could not seek to start, WAV size headers not updated!\n");
 180                 else if (data_length > 0x7ffff000)
 181                         mp_msg(MSGT_AO, MSGL_ERR, "File larger than allowed for WAV files, may play truncated!\n");
 182                 else {
 183                 wavhdr.file_length = data_length + sizeof(wavhdr) - 8;
 184                 wavhdr.file_length = le2me_32(wavhdr.file_length);
 185                 wavhdr.data_length = le2me_32(data_length);
 186                 fwrite(&wavhdr,sizeof(wavhdr),1,fp);
 187                 }
 188         }
 189         fclose(fp);
 190         if (ao_outputfilename)
 191           free(ao_outputfilename);
 192         ao_outputfilename = NULL;
 193 }
 194
 195 // stop playing and empty buffers (for seeking/pause)
 196 static void reset(void){
 197
 198 }
 199
 200 // stop playing, keep buffers (for pause)
 201 static void audio_pause(void)
 202 {
 203     // for now, just call reset();
 204     reset();
 205 }
 206
 207 // resume playing, after audio_pause()
 208 static void audio_resume(void)
 209 {
 210 }
 211
 212 // return: how many bytes can be played without blocking
 213 static int get_space(void){
 214
 215     if(vo_pts)
 216       return ao_data.pts < vo_pts + fast * 30000 ? ao_data.outburst : 0;
 217     return ao_data.outburst;
 218 }
 219
 220 // plays 'len' bytes of 'data'
 221 // it should round it down to outburst*n
 222 // return: number of bytes played
 223 static int play(void* data,int len,int flags){
 224
 225 // let libaf to do the conversion...
 226 #if 0
 227 //#ifdef WORDS_BIGENDIAN
 228         if (ao_data.format == AFMT_S16_LE) {
 229           unsigned short *buffer = (unsigned short *) data;
 230           register int i;
 231           for(i = 0; i < len/2; ++i) {
 232             buffer[i] = le2me_16(buffer[i]);
 233           }
 234         }
 235 #endif
 236
 237         if (ao_data.channels == 6 || ao_data.channels == 5) {
 238                 int frame_size = le2me_16(wavhdr.bits) / 8;
 239                 len -= len % (frame_size * ao_data.channels);
 240                 reorder_channel_nch(data, AF_CHANNEL_LAYOUT_MPLAYER_DEFAULT,
 241                                     AF_CHANNEL_LAYOUT_WAVEEX_DEFAULT,
 242                                     ao_data.channels,
 243                                     len / frame_size, frame_size);
 244         }
 245
 246         //printf("PCM: Writing chunk!\n");
 247         fwrite(data,len,1,fp);
 248
 249         if(ao_pcm_waveheader)
 250                 data_length += len;
 251
 252         return len;
 253 }
 254
 255 // return: delay in seconds between first and last sample in buffer
 256 static float get_delay(void){
 257
 258     return 0.0;
 259 }
 260
 261
 262
 263
 264
 265