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
  36 #ifdef __MINGW32__
  37 // for GetFileType to detect pipes
  38 #include <windows.h>
  39 #endif
  40
  41 static const ao_info_t info =
  42 {
  43     "RAW PCM/WAVE file writer audio output",
  44     "pcm",
  45     "Atmosfear",
  46     ""
  47 };
  48
  49 LIBAO_EXTERN(pcm)
  50
  51 extern int vo_pts;
  52
  53 static char *ao_outputfilename = NULL;
  54 static int ao_pcm_waveheader = 1;
  55 static int fast = 0;
  56
  57 #define WAV_ID_RIFF 0x46464952 /* "RIFF" */
  58 #define WAV_ID_WAVE 0x45564157 /* "WAVE" */
  59 #define WAV_ID_FMT  0x20746d66 /* "fmt " */
  60 #define WAV_ID_DATA 0x61746164 /* "data" */
  61 #define WAV_ID_PCM  0x0001
  62 #define WAV_ID_FLOAT_PCM  0x0003
  63 #define WAV_ID_FORMAT_EXTENSIBLE 0xfffe
  64
  65 /* init with default values */
  66 static uint64_t data_length;
  67 static FILE *fp = NULL;
  68
  69
  70 static void fput16le(uint16_t val, FILE *fp) {
  71     uint8_t bytes[2] = {val, val >> 8};
  72     fwrite(bytes, 1, 2, fp);
  73 }
  74
  75 static void fput32le(uint32_t val, FILE *fp) {
  76     uint8_t bytes[4] = {val, val >> 8, val >> 16, val >> 24};
  77     fwrite(bytes, 1, 4, fp);
  78 }
  79
  80 static void write_wave_header(FILE *fp, uint64_t data_length) {
  81     int use_waveex = (ao_data.channels >= 5 && ao_data.channels <= 8);
  82     uint16_t fmt = (ao_data.format == AF_FORMAT_FLOAT_LE) ? WAV_ID_FLOAT_PCM : WAV_ID_PCM;
  83     uint32_t fmt_chunk_size = use_waveex ? 40 : 16;
  84     int bits = af_fmt2bits(ao_data.format);
  85
  86     // Master RIFF chunk
  87     fput32le(WAV_ID_RIFF, fp);
  88     // RIFF chunk size: 'WAVE' + 'fmt ' + 4 + fmt_chunk_size + data chunk hdr (8) + data length
  89     fput32le(12 + fmt_chunk_size + 8 + data_length, fp);
  90     fput32le(WAV_ID_WAVE, fp);
  91
  92     // Format chunk
  93     fput32le(WAV_ID_FMT, fp);
  94     fput32le(fmt_chunk_size, fp);
  95     fput16le(use_waveex ? WAV_ID_FORMAT_EXTENSIBLE : fmt, fp);
  96     fput16le(ao_data.channels, fp);
  97     fput32le(ao_data.samplerate, fp);
  98     fput32le(ao_data.bps, fp);
  99     fput16le(ao_data.channels * (bits / 8), fp);
 100     fput16le(bits, fp);
 101
 102     if (use_waveex) {
 103         // Extension chunk
 104         fput16le(22, fp);
 105         fput16le(bits, fp);
 106         switch (ao_data.channels) {
 107             case 5:
 108                 fput32le(0x0607, fp); // L R C Lb Rb
 109                 break;
 110             case 6:
 111                 fput32le(0x060f, fp); // L R C Lb Rb LFE
 112                 break;
 113             case 7:
 114                 fput32le(0x0727, fp); // L R C Cb Ls Rs LFE
 115                 break;
 116             case 8:
 117                 fput32le(0x063f, fp); // L R C Lb Rb Ls Rs LFE
 118                 break;
 119         }
 120         // 2 bytes format + 14 bytes guid
 121         fput32le(fmt, fp);
 122         fput32le(0x00100000, fp);
 123         fput32le(0xAA000080, fp);
 124         fput32le(0x719B3800, fp);
 125     }
 126
 127     // Data chunk
 128     fput32le(WAV_ID_DATA, fp);
 129     fput32le(data_length, fp);
 130 }
 131
 132 // to set/get/query special features/parameters
 133 static int control(int cmd,void *arg){
 134     return -1;
 135 }
 136
 137 // open & setup audio device
 138 // return: 1=success 0=fail
 139 static int init(int rate,int channels,int format,int flags){
 140     const opt_t subopts[] = {
 141         {"waveheader", OPT_ARG_BOOL, &ao_pcm_waveheader, NULL},
 142         {"file",       OPT_ARG_MSTRZ, &ao_outputfilename, NULL},
 143         {"fast",       OPT_ARG_BOOL, &fast, NULL},
 144         {NULL}
 145     };
 146     // set defaults
 147     ao_pcm_waveheader = 1;
 148
 149     if (subopt_parse(ao_subdevice, subopts) != 0) {
 150         return 0;
 151     }
 152     if (!ao_outputfilename){
 153         ao_outputfilename =
 154             strdup(ao_pcm_waveheader?"audiodump.wav":"audiodump.pcm");
 155     }
 156
 157     if (ao_pcm_waveheader)
 158     {
 159         // WAV files must have one of the following formats
 160
 161         switch(format){
 162         case AF_FORMAT_U8:
 163         case AF_FORMAT_S16_LE:
 164         case AF_FORMAT_S24_LE:
 165         case AF_FORMAT_S32_LE:
 166         case AF_FORMAT_FLOAT_LE:
 167         case AF_FORMAT_AC3_BE:
 168         case AF_FORMAT_AC3_LE:
 169              break;
 170         default:
 171             format = AF_FORMAT_S16_LE;
 172             break;
 173         }
 174     }
 175
 176     ao_data.outburst = 65536;
 177     ao_data.buffersize= 2*65536;
 178     ao_data.channels=channels;
 179     ao_data.samplerate=rate;
 180     ao_data.format=format;
 181     ao_data.bps=channels*rate*(af_fmt2bits(format)/8);
 182
 183     mp_tmsg(MSGT_AO, MSGL_INFO, "[AO PCM] File: %s (%s)\nPCM: Samplerate: %iHz Channels: %s Format %s\n", ao_outputfilename,
 184            (ao_pcm_waveheader?"WAVE":"RAW PCM"), rate,
 185            (channels > 1) ? "Stereo" : "Mono", af_fmt2str_short(format));
 186     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");
 187
 188     fp = fopen(ao_outputfilename, "wb");
 189     if(fp) {
 190         if(ao_pcm_waveheader){ /* Reserve space for wave header */
 191             write_wave_header(fp, 0x7ffff000);
 192         }
 193         return 1;
 194     }
 195     mp_tmsg(MSGT_AO, MSGL_ERR, "[AO PCM] Failed to open %s for writing!\n",
 196                ao_outputfilename);
 197     return 0;
 198 }
 199
 200 // close audio device
 201 static void uninit(int immed){
 202
 203     if(ao_pcm_waveheader){ /* Rewrite wave header */
 204         int broken_seek = 0;
 205 #ifdef __MINGW32__
 206         // Windows, in its usual idiocy "emulates" seeks on pipes so it always looks
 207         // like they work. So we have to detect them brute-force.
 208         broken_seek = GetFileType((HANDLE)_get_osfhandle(_fileno(fp))) != FILE_TYPE_DISK;
 209 #endif
 210         if (broken_seek || fseek(fp, 0, SEEK_SET) != 0)
 211             mp_msg(MSGT_AO, MSGL_ERR, "Could not seek to start, WAV size headers not updated!\n");
 212         else {
 213             if (data_length > 0xfffff000) {
 214                 mp_msg(MSGT_AO, MSGL_ERR, "File larger than allowed for WAV files, may play truncated!\n");
 215                 data_length = 0xfffff000;
 216             }
 217             write_wave_header(fp, data_length);
 218         }
 219     }
 220     fclose(fp);
 221     if (ao_outputfilename)
 222         free(ao_outputfilename);
 223     ao_outputfilename = NULL;
 224 }
 225
 226 // stop playing and empty buffers (for seeking/pause)
 227 static void reset(void){
 228
 229 }
 230
 231 // stop playing, keep buffers (for pause)
 232 static void audio_pause(void)
 233 {
 234     // for now, just call reset();
 235     reset();
 236 }
 237
 238 // resume playing, after audio_pause()
 239 static void audio_resume(void)
 240 {
 241 }
 242
 243 // return: how many bytes can be played without blocking
 244 static int get_space(void){
 245
 246     if(vo_pts)
 247         return ao_data.pts < vo_pts + fast * 30000 ? ao_data.outburst : 0;
 248     return ao_data.outburst;
 249 }
 250
 251 // plays 'len' bytes of 'data'
 252 // it should round it down to outburst*n
 253 // return: number of bytes played
 254 static int play(void* data,int len,int flags){
 255
 256 // let libaf to do the conversion...
 257 #if 0
 258 //#if HAVE_BIGENDIAN
 259     if (ao_data.format == AFMT_S16_LE) {
 260       unsigned short *buffer = (unsigned short *) data;
 261       register int i;
 262       for(i = 0; i < len/2; ++i) {
 263         buffer[i] = le2me_16(buffer[i]);
 264       }
 265     }
 266 #endif
 267
 268     if (ao_data.channels == 5 || ao_data.channels == 6 || ao_data.channels == 8) {
 269         int frame_size = af_fmt2bits(ao_data.format) / 8;
 270         len -= len % (frame_size * ao_data.channels);
 271         reorder_channel_nch(data, AF_CHANNEL_LAYOUT_MPLAYER_DEFAULT,
 272                             AF_CHANNEL_LAYOUT_WAVEEX_DEFAULT,
 273                             ao_data.channels,
 274                             len / frame_size, frame_size);
 275     }
 276
 277     //printf("PCM: Writing chunk!\n");
 278     fwrite(data,len,1,fp);
 279
 280     if(ao_pcm_waveheader)
 281         data_length += len;
 282
 283     return len;
 284 }
 285
 286 // return: delay in seconds between first and last sample in buffer
 287 static float get_delay(void){
 288
 289     return 0.0;
 290 }