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 #ifdef __MINGW32__
  38 // for GetFileType to detect pipes
  39 #include <windows.h>
  40 #endif
  41
  42 static const ao_info_t info =
  43 {
  44     "RAW PCM/WAVE file writer audio output",
  45     "pcm",
  46     "Atmosfear",
  47     ""
  48 };
  49
  50 LIBAO_EXTERN(pcm)
  51
  52 extern int vo_pts;
  53
  54 static char *ao_outputfilename = NULL;
  55 static int ao_pcm_waveheader = 1;
  56 static int fast = 0;
  57
  58 #define WAV_ID_RIFF 0x46464952 /* "RIFF" */
  59 #define WAV_ID_WAVE 0x45564157 /* "WAVE" */
  60 #define WAV_ID_FMT  0x20746d66 /* "fmt " */
  61 #define WAV_ID_DATA 0x61746164 /* "data" */
  62 #define WAV_ID_PCM  0x0001
  63 #define WAV_ID_FLOAT_PCM  0x0003
  64
  65 struct WaveHeader
  66 {
  67     uint32_t riff;
  68     uint32_t file_length;
  69     uint32_t wave;
  70     uint32_t fmt;
  71     uint32_t fmt_length;
  72     uint16_t fmt_tag;
  73     uint16_t channels;
  74     uint32_t sample_rate;
  75     uint32_t bytes_per_second;
  76     uint16_t block_align;
  77     uint16_t bits;
  78     uint32_t data;
  79     uint32_t data_length;
  80 };
  81
  82 /* init with default values */
  83 static struct WaveHeader wavhdr;
  84 static uint64_t data_length;
  85
  86 static FILE *fp = NULL;
  87
  88 // to set/get/query special features/parameters
  89 static int control(int cmd,void *arg){
  90     return -1;
  91 }
  92
  93 // open & setup audio device
  94 // return: 1=success 0=fail
  95 static int init(int rate,int channels,int format,int flags){
  96     int bits;
  97     const opt_t subopts[] = {
  98         {"waveheader", OPT_ARG_BOOL, &ao_pcm_waveheader, NULL},
  99         {"file",       OPT_ARG_MSTRZ, &ao_outputfilename, NULL},
 100         {"fast",       OPT_ARG_BOOL, &fast, NULL},
 101         {NULL}
 102     };
 103     // set defaults
 104     ao_pcm_waveheader = 1;
 105
 106     if (subopt_parse(ao_subdevice, subopts) != 0) {
 107         return 0;
 108     }
 109     if (!ao_outputfilename){
 110         ao_outputfilename =
 111             strdup(ao_pcm_waveheader?"audiodump.wav":"audiodump.pcm");
 112     }
 113
 114     bits=8;
 115     switch(format){
 116     case AF_FORMAT_S32_BE:
 117         format=AF_FORMAT_S32_LE;
 118     case AF_FORMAT_S32_LE:
 119         bits=32;
 120         break;
 121     case AF_FORMAT_FLOAT_BE:
 122         format=AF_FORMAT_FLOAT_LE;
 123     case AF_FORMAT_FLOAT_LE:
 124         bits=32;
 125         break;
 126     case AF_FORMAT_S8:
 127         format=AF_FORMAT_U8;
 128     case AF_FORMAT_U8:
 129         break;
 130     case AF_FORMAT_AC3:
 131         bits=16;
 132         break;
 133     default:
 134         format=AF_FORMAT_S16_LE;
 135         bits=16;
 136         break;
 137     }
 138
 139     ao_data.outburst = 65536;
 140     ao_data.buffersize= 2*65536;
 141     ao_data.channels=channels;
 142     ao_data.samplerate=rate;
 143     ao_data.format=format;
 144     ao_data.bps=channels*rate*(bits/8);
 145
 146     wavhdr.riff = le2me_32(WAV_ID_RIFF);
 147     wavhdr.wave = le2me_32(WAV_ID_WAVE);
 148     wavhdr.fmt = le2me_32(WAV_ID_FMT);
 149     wavhdr.fmt_length = le2me_32(16);
 150     wavhdr.fmt_tag = le2me_16(format == AF_FORMAT_FLOAT_LE ? WAV_ID_FLOAT_PCM : WAV_ID_PCM);
 151     wavhdr.channels = le2me_16(ao_data.channels);
 152     wavhdr.sample_rate = le2me_32(ao_data.samplerate);
 153     wavhdr.bytes_per_second = le2me_32(ao_data.bps);
 154     wavhdr.bits = le2me_16(bits);
 155     wavhdr.block_align = le2me_16(ao_data.channels * (bits / 8));
 156
 157     wavhdr.data = le2me_32(WAV_ID_DATA);
 158     wavhdr.data_length=le2me_32(0x7ffff000);
 159     wavhdr.file_length = wavhdr.data_length + sizeof(wavhdr) - 8;
 160
 161     mp_tmsg(MSGT_AO, MSGL_INFO, "[AO PCM] File: %s (%s)\nPCM: Samplerate: %iHz Channels: %s Format %s\n", ao_outputfilename,
 162            (ao_pcm_waveheader?"WAVE":"RAW PCM"), rate,
 163            (channels > 1) ? "Stereo" : "Mono", af_fmt2str_short(format));
 164     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");
 165
 166     fp = fopen(ao_outputfilename, "wb");
 167     if(fp) {
 168         if(ao_pcm_waveheader){ /* Reserve space for wave header */
 169             fwrite(&wavhdr,sizeof(wavhdr),1,fp);
 170         }
 171         return 1;
 172     }
 173     mp_tmsg(MSGT_AO, MSGL_ERR, "[AO PCM] Failed to open %s for writing!\n",
 174                ao_outputfilename);
 175     return 0;
 176 }
 177
 178 // close audio device
 179 static void uninit(int immed){
 180
 181     if(ao_pcm_waveheader){ /* Rewrite wave header */
 182         int broken_seek = 0;
 183 #ifdef __MINGW32__
 184         // Windows, in its usual idiocy "emulates" seeks on pipes so it always looks
 185         // like they work. So we have to detect them brute-force.
 186         broken_seek = GetFileType((HANDLE)_get_osfhandle(_fileno(fp))) != FILE_TYPE_DISK;
 187 #endif
 188         if (broken_seek || fseek(fp, 0, SEEK_SET) != 0)
 189             mp_msg(MSGT_AO, MSGL_ERR, "Could not seek to start, WAV size headers not updated!\n");
 190         else if (data_length > 0x7ffff000)
 191             mp_msg(MSGT_AO, MSGL_ERR, "File larger than allowed for WAV files, may play truncated!\n");
 192         else {
 193             wavhdr.file_length = data_length + sizeof(wavhdr) - 8;
 194             wavhdr.file_length = le2me_32(wavhdr.file_length);
 195             wavhdr.data_length = le2me_32(data_length);
 196             fwrite(&wavhdr,sizeof(wavhdr),1,fp);
 197         }
 198     }
 199     fclose(fp);
 200     if (ao_outputfilename)
 201         free(ao_outputfilename);
 202     ao_outputfilename = NULL;
 203 }
 204
 205 // stop playing and empty buffers (for seeking/pause)
 206 static void reset(void){
 207
 208 }
 209
 210 // stop playing, keep buffers (for pause)
 211 static void audio_pause(void)
 212 {
 213     // for now, just call reset();
 214     reset();
 215 }
 216
 217 // resume playing, after audio_pause()
 218 static void audio_resume(void)
 219 {
 220 }
 221
 222 // return: how many bytes can be played without blocking
 223 static int get_space(void){
 224
 225     if(vo_pts)
 226         return ao_data.pts < vo_pts + fast * 30000 ? ao_data.outburst : 0;
 227     return ao_data.outburst;
 228 }
 229
 230 // plays 'len' bytes of 'data'
 231 // it should round it down to outburst*n
 232 // return: number of bytes played
 233 static int play(void* data,int len,int flags){
 234
 235 // let libaf to do the conversion...
 236 #if 0
 237 //#if HAVE_BIGENDIAN
 238     if (ao_data.format == AFMT_S16_LE) {
 239       unsigned short *buffer = (unsigned short *) data;
 240       register int i;
 241       for(i = 0; i < len/2; ++i) {
 242         buffer[i] = le2me_16(buffer[i]);
 243       }
 244     }
 245 #endif
 246
 247     if (ao_data.channels == 5 || ao_data.channels == 6 || ao_data.channels == 8) {
 248         int frame_size = le2me_16(wavhdr.bits) / 8;
 249         len -= len % (frame_size * ao_data.channels);
 250         reorder_channel_nch(data, AF_CHANNEL_LAYOUT_MPLAYER_DEFAULT,
 251                             AF_CHANNEL_LAYOUT_WAVEEX_DEFAULT,
 252                             ao_data.channels,
 253                             len / frame_size, frame_size);
 254     }
 255
 256     //printf("PCM: Writing chunk!\n");
 257     fwrite(data,len,1,fp);
 258
 259     if(ao_pcm_waveheader)
 260         data_length += len;
 261
 262     return len;
 263 }
 264
 265 // return: delay in seconds between first and last sample in buffer
 266 static float get_delay(void){
 267
 268     return 0.0;
 269 }