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 static 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