apps/plugins/mpegplayer/audio_thread.c

   1 /***************************************************************************
   2  *             __________               __   ___.
   3  *   Open      \______   \ ____   ____ |  | _\_ |__   _______  ___
   4  *   Source     |       _//  _ \_/ ___\|  |/ /| __ \ /  _ \  \/  /
   5  *   Jukebox    |    |   (  <_> )  \___|    < | \_\ (  <_> > <  <
   6  *   Firmware   |____|_  /\____/ \___  >__|_ \|___  /\____/__/\_ \
   7  *                     \/            \/     \/    \/            \/
   8  * $Id$
   9  *
  10  * mpegplayer audio thread implementation
  11  *
  12  * Copyright (c) 2007 Michael Sevakis
  13  *
  14  * This program is free software; you can redistribute it and/or
  15  * modify it under the terms of the GNU General Public License
  16  * as published by the Free Software Foundation; either version 2
  17  * of the License, or (at your option) any later version.
  18  *
  19  * This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
  20  * KIND, either express or implied.
  21  *
  22  ****************************************************************************/
  23 #include "plugin.h"
  24 #include "mpegplayer.h"
  25 #include "codecs/libmad/bit.h"
  26 #include "codecs/libmad/mad.h"
  27
  28 /** Audio stream and thread **/
  29 struct pts_queue_slot;
  30 struct audio_thread_data
  31 {
  32     struct queue_event ev;  /* Our event queue to receive commands */
  33     int state;              /* Thread state */
  34     int status;             /* Media status (STREAM_PLAYING, etc.) */
  35     int mad_errors;         /* A count of the errors in each frame */
  36     unsigned samplerate;    /* Current stream sample rate */
  37     int nchannels;          /* Number of audio channels */
  38     struct dsp_config *dsp; /* The DSP we're using */
  39 };
  40
  41 /* The audio thread is stolen from the core codec thread */
  42 static struct event_queue audio_str_queue SHAREDBSS_ATTR;
  43 static struct queue_sender_list audio_str_queue_send SHAREDBSS_ATTR;
  44 struct stream audio_str IBSS_ATTR;
  45
  46 /* libmad related definitions */
  47 static struct mad_stream stream IBSS_ATTR;
  48 static struct mad_frame  frame IBSS_ATTR;
  49 static struct mad_synth  synth IBSS_ATTR;
  50
  51 /*sbsample buffer for mad_frame*/
  52 mad_fixed_t sbsample[2][36][32];
  53
  54 /* 2567 bytes */
  55 static unsigned char mad_main_data[MAD_BUFFER_MDLEN];
  56
  57 /* There isn't enough room for this in IRAM on PortalPlayer, but there
  58    is for Coldfire. */
  59
  60 /* 4608 bytes */
  61 #if defined(CPU_COLDFIRE) || defined(CPU_S5L870X)
  62 static mad_fixed_t mad_frame_overlap[2][32][18] IBSS_ATTR;
  63 #else
  64 static mad_fixed_t mad_frame_overlap[2][32][18];
  65 #endif
  66
  67 /** A queue for saving needed information about MPEG audio packets **/
  68 #define AUDIODESC_QUEUE_LEN  (1 << 5) /* 32 should be way more than sufficient -
  69                                          if not, the case is handled */
  70 #define AUDIODESC_QUEUE_MASK (AUDIODESC_QUEUE_LEN-1)
  71 struct audio_frame_desc
  72 {
  73     uint32_t time;  /* Time stamp for packet in audio ticks       */
  74     ssize_t  size;  /* Number of unprocessed bytes left in packet */
  75 };
  76
  77  /* This starts out wr == rd but will never be emptied to zero during
  78     streaming again in order to support initializing the first packet's
  79     timestamp without a special case */
  80 struct
  81 {
  82     /* Compressed audio data */
  83     uint8_t *start;  /* Start of encoded audio buffer */
  84     uint8_t *ptr;    /* Pointer to next encoded audio data */
  85     ssize_t used;    /* Number of bytes in MPEG audio buffer */
  86     /* Compressed audio data descriptors */
  87     unsigned read, write;
  88     struct audio_frame_desc *curr; /* Current slot */
  89     struct audio_frame_desc descs[AUDIODESC_QUEUE_LEN];
  90 } audio_queue;
  91
  92 static inline int audiodesc_queue_count(void)
  93 {
  94     return audio_queue.write - audio_queue.read;
  95 }
  96
  97 static inline bool audiodesc_queue_full(void)
  98 {
  99     return audio_queue.used >= MPA_MAX_FRAME_SIZE + MAD_BUFFER_GUARD ||
 100             audiodesc_queue_count() >= AUDIODESC_QUEUE_LEN;
 101 }
 102
 103 /* Increments the queue tail postion - should be used to preincrement */
 104 static inline void audiodesc_queue_add_tail(void)
 105 {
 106     if (audiodesc_queue_full())
 107     {
 108         DEBUGF("audiodesc_queue_add_tail: audiodesc queue full!\n");
 109         return;
 110     }
 111
 112     audio_queue.write++;
 113 }
 114
 115 /* Increments the queue tail position - leaves one slot as current */
 116 static inline bool audiodesc_queue_remove_head(void)
 117 {
 118     if (audio_queue.write == audio_queue.read)
 119         return false;
 120
 121     audio_queue.read++;
 122     return true;
 123 }
 124
 125 /* Returns the "tail" at the index just behind the write index */
 126 static inline struct audio_frame_desc * audiodesc_queue_tail(void)
 127 {
 128     return &audio_queue.descs[(audio_queue.write - 1) & AUDIODESC_QUEUE_MASK];
 129 }
 130
 131 /* Returns a pointer to the current head */
 132 static inline struct audio_frame_desc * audiodesc_queue_head(void)
 133 {
 134     return &audio_queue.descs[audio_queue.read & AUDIODESC_QUEUE_MASK];
 135 }
 136
 137 /* Resets the pts queue - call when starting and seeking */
 138 static void audio_queue_reset(void)
 139 {
 140     audio_queue.ptr = audio_queue.start;
 141     audio_queue.used = 0;
 142     audio_queue.read = 0;
 143     audio_queue.write = 0;
 144     rb->memset(audio_queue.descs, 0, sizeof (audio_queue.descs));
 145     audio_queue.curr = audiodesc_queue_head();
 146 }
 147
 148 static void audio_queue_advance_pos(ssize_t len)
 149 {
 150     audio_queue.ptr        += len;
 151     audio_queue.used       -= len;
 152     audio_queue.curr->size -= len;
 153 }
 154
 155 static int audio_buffer(struct stream *str, enum stream_parse_mode type)
 156 {
 157     int ret = STREAM_OK;
 158
 159     /* Carry any overshoot to the next size since we're technically
 160        -size bytes into it already. If size is negative an audio
 161        frame was split across packets. Old has to be saved before
 162        moving the head. */
 163     if (audio_queue.curr->size <= 0 && audiodesc_queue_remove_head())
 164     {
 165         struct audio_frame_desc *old = audio_queue.curr;
 166         audio_queue.curr = audiodesc_queue_head();
 167         audio_queue.curr->size += old->size;
 168         old->size = 0;
 169     }
 170
 171     /* Add packets to compressed audio buffer until it's full or the
 172      * timestamp queue is full - whichever happens first */
 173     while (!audiodesc_queue_full())
 174     {
 175         ret = parser_get_next_data(str, type);
 176         struct audio_frame_desc *curr;
 177         ssize_t len;
 178
 179         if (ret != STREAM_OK)
 180             break;
 181
 182         /* Get data from next audio packet */
 183         len = str->curr_packet_end - str->curr_packet;
 184
 185         if (str->pkt_flags & PKT_HAS_TS)
 186         {
 187             audiodesc_queue_add_tail();
 188             curr = audiodesc_queue_tail();
 189             curr->time = TS_TO_TICKS(str->pts);
 190             /* pts->size should have been zeroed when slot was
 191                freed */
 192         }
 193         else
 194         {
 195             /* Add to the one just behind the tail - this may be
 196              * the head or the previouly added tail - whether or
 197              * not we'll ever reach this is quite in question
 198              * since audio always seems to have every packet
 199              * timestamped */
 200             curr = audiodesc_queue_tail();
 201         }
 202
 203         curr->size += len;
 204
 205         /* Slide any remainder over to beginning */
 206         if (audio_queue.ptr > audio_queue.start && audio_queue.used > 0)
 207         {
 208             rb->memmove(audio_queue.start, audio_queue.ptr,
 209                         audio_queue.used);
 210         }
 211
 212         /* Splice this packet onto any remainder */
 213         rb->memcpy(audio_queue.start + audio_queue.used,
 214                    str->curr_packet, len);
 215
 216         audio_queue.used += len;
 217         audio_queue.ptr = audio_queue.start;
 218
 219         rb->yield();
 220     }
 221
 222     return ret;
 223 }
 224
 225 /* Initialise libmad */
 226 static void init_mad(void)
 227 {
 228     /*init the sbsample buffer*/
 229     frame.sbsample = &sbsample;
 230     frame.sbsample_prev = &sbsample;
 231
 232     mad_stream_init(&stream);
 233     mad_frame_init(&frame);
 234     mad_synth_init(&synth);
 235
 236     /* We do this so libmad doesn't try to call codec_calloc() */
 237     rb->memset(mad_frame_overlap, 0, sizeof(mad_frame_overlap));
 238     frame.overlap = (void *)mad_frame_overlap;
 239
 240     rb->memset(mad_main_data, 0, sizeof(mad_main_data));
 241     stream.main_data = &mad_main_data;
 242 }
 243
 244 /* Sync audio stream to a particular frame - see main decoder loop for
 245  * detailed remarks */
 246 static int audio_sync(struct audio_thread_data *td,
 247                       struct str_sync_data *sd)
 248 {
 249     int retval = STREAM_MATCH;
 250     uint32_t sdtime = TS_TO_TICKS(clip_time(&audio_str, sd->time));
 251     uint32_t time;
 252     uint32_t duration = 0;
 253     struct stream *str;
 254     struct stream tmp_str;
 255     struct mad_header header;
 256     struct mad_stream stream;
 257
 258     if (td->ev.id == STREAM_SYNC)
 259     {
 260         /* Actually syncing for playback - use real stream */
 261         time = 0;
 262         str = &audio_str;
 263     }
 264     else
 265     {
 266         /* Probing - use temp stream */
 267         time = INVALID_TIMESTAMP;
 268         str = &tmp_str;
 269         str->id = audio_str.id;
 270     }
 271
 272     str->hdr.pos = sd->sk.pos;
 273     str->hdr.limit = sd->sk.pos + sd->sk.len;
 274
 275     mad_stream_init(&stream);
 276     mad_header_init(&header);
 277
 278     while (1)
 279     {
 280         if (audio_buffer(str, STREAM_PM_RANDOM_ACCESS) == STREAM_DATA_END)
 281         {
 282             DEBUGF("audio_sync:STR_DATA_END\n  aqu:%zd swl:%ld swr:%ld\n",
 283                     audio_queue.used, str->hdr.win_left, str->hdr.win_right);
 284             if (audio_queue.used <= MAD_BUFFER_GUARD)
 285                 goto sync_data_end;
 286         }
 287
 288         stream.error = 0;
 289         mad_stream_buffer(&stream, audio_queue.ptr, audio_queue.used);
 290
 291         if (stream.sync && mad_stream_sync(&stream) < 0)
 292         {
 293             DEBUGF(" audio: mad_stream_sync failed\n");
 294             audio_queue_advance_pos(MAX(audio_queue.curr->size - 1, 1));
 295             continue;
 296         }
 297
 298         stream.sync = 0;
 299
 300         if (mad_header_decode(&header, &stream) < 0)
 301         {
 302             DEBUGF(" audio: mad_header_decode failed:%s\n",
 303                    mad_stream_errorstr(&stream));
 304             audio_queue_advance_pos(1);
 305             continue;
 306         }
 307
 308         duration = 32*MAD_NSBSAMPLES(&header);
 309         time = audio_queue.curr->time;
 310
 311         DEBUGF(" audio: ft:%u t:%u fe:%u nsamp:%u sampr:%u\n",
 312                (unsigned)TICKS_TO_TS(time), (unsigned)sd->time,
 313                (unsigned)TICKS_TO_TS(time + duration),
 314                (unsigned)duration, header.samplerate);
 315
 316         audio_queue_advance_pos(stream.this_frame - audio_queue.ptr);
 317
 318         if (time <= sdtime && sdtime < time + duration)
 319         {
 320             DEBUGF(" audio: ft<=t<fe\n");
 321             retval = STREAM_PERFECT_MATCH;
 322             break;
 323         }
 324         else if (time > sdtime)
 325         {
 326             DEBUGF(" audio: ft>t\n");
 327             break;
 328         }
 329
 330         audio_queue_advance_pos(stream.next_frame - audio_queue.ptr);
 331         audio_queue.curr->time += duration;
 332
 333         rb->yield();
 334     }
 335
 336 sync_data_end:
 337     if (td->ev.id == STREAM_FIND_END_TIME)
 338     {
 339         if (time != INVALID_TIMESTAMP)
 340         {
 341             time = TICKS_TO_TS(time);
 342             duration = TICKS_TO_TS(duration);
 343             sd->time = time + duration;
 344             retval = STREAM_PERFECT_MATCH;
 345         }
 346         else
 347         {
 348             retval = STREAM_NOT_FOUND;
 349         }
 350     }
 351
 352     DEBUGF(" audio header: 0x%02X%02X%02X%02X\n",
 353            (unsigned)audio_queue.ptr[0], (unsigned)audio_queue.ptr[1],
 354            (unsigned)audio_queue.ptr[2], (unsigned)audio_queue.ptr[3]);
 355
 356     return retval;
 357     (void)td;
 358 }
 359
 360 static void audio_thread_msg(struct audio_thread_data *td)
 361 {
 362     while (1)
 363     {
 364         intptr_t reply = 0;
 365
 366         switch (td->ev.id)
 367         {
 368         case STREAM_PLAY:
 369             td->status = STREAM_PLAYING;
 370
 371             switch (td->state)
 372             {
 373             case TSTATE_INIT:
 374                 td->state = TSTATE_DECODE;
 375             case TSTATE_DECODE:
 376             case TSTATE_RENDER_WAIT:
 377             case TSTATE_RENDER_WAIT_END:
 378                 break;
 379
 380             case TSTATE_EOS:
 381                 /* At end of stream - no playback possible so fire the
 382                  * completion event */
 383                 stream_generate_event(&audio_str, STREAM_EV_COMPLETE, 0);
 384                 break;
 385             }
 386
 387             break;
 388
 389         case STREAM_PAUSE:
 390             td->status = STREAM_PAUSED;
 391             reply = td->state != TSTATE_EOS;
 392             break;
 393
 394         case STREAM_STOP:
 395             if (td->state == TSTATE_DATA)
 396                 stream_clear_notify(&audio_str, DISK_BUF_DATA_NOTIFY);
 397
 398             td->status = STREAM_STOPPED;
 399             td->state = TSTATE_EOS;
 400
 401             reply = true;
 402             break;
 403
 404         case STREAM_RESET:
 405             if (td->state == TSTATE_DATA)
 406                 stream_clear_notify(&audio_str, DISK_BUF_DATA_NOTIFY);
 407
 408             td->status = STREAM_STOPPED;
 409             td->state = TSTATE_INIT;
 410             td->samplerate = 0;
 411             td->nchannels = 0;
 412
 413             init_mad();
 414             td->mad_errors = 0;
 415
 416             audio_queue_reset();
 417
 418             reply = true;
 419             break;
 420
 421         case STREAM_NEEDS_SYNC:
 422             reply = true; /* Audio always needs to */
 423             break;
 424
 425         case STREAM_SYNC:
 426         case STREAM_FIND_END_TIME:
 427             if (td->state != TSTATE_INIT)
 428                 break;
 429
 430             reply = audio_sync(td, (struct str_sync_data *)td->ev.data);
 431             break;
 432
 433         case DISK_BUF_DATA_NOTIFY:
 434             /* Our bun is done */
 435             if (td->state != TSTATE_DATA)
 436                 break;
 437
 438             td->state = TSTATE_DECODE;
 439             str_data_notify_received(&audio_str);
 440             break;
 441
 442         case STREAM_QUIT:
 443             /* Time to go - make thread exit */
 444             td->state = TSTATE_EOS;
 445             return;
 446         }
 447
 448         str_reply_msg(&audio_str, reply);
 449
 450         if (td->status == STREAM_PLAYING)
 451         {
 452             switch (td->state)
 453             {
 454             case TSTATE_DECODE:
 455             case TSTATE_RENDER_WAIT:
 456             case TSTATE_RENDER_WAIT_END:
 457                 /* These return when in playing state */
 458                 return;
 459             }
 460         }
 461
 462         str_get_msg(&audio_str, &td->ev);
 463     }
 464 }
 465
 466 static void audio_thread(void)
 467 {
 468     struct audio_thread_data td;
 469 #ifdef HAVE_PRIORITY_SCHEDULING
 470     /* Up the priority since the core DSP over-yields internally */
 471     int old_priority = rb->thread_set_priority(THREAD_ID_CURRENT,
 472                                                PRIORITY_PLAYBACK-4);
 473 #endif
 474
 475     rb->memset(&td, 0, sizeof (td));
 476     td.status = STREAM_STOPPED;
 477     td.state = TSTATE_EOS;
 478
 479     /* We need this here to init the EMAC for Coldfire targets */
 480     init_mad();
 481
 482     td.dsp = (struct dsp_config *)rb->dsp_configure(NULL, DSP_MYDSP,
 483                                                     CODEC_IDX_AUDIO);
 484     rb->sound_set_pitch(PITCH_SPEED_100);
 485     rb->dsp_configure(td.dsp, DSP_RESET, 0);
 486     rb->dsp_configure(td.dsp, DSP_SET_SAMPLE_DEPTH, MAD_F_FRACBITS);
 487
 488     goto message_wait;
 489
 490     /* This is the decoding loop. */
 491     while (1)
 492     {
 493         td.state = TSTATE_DECODE;
 494
 495         /* Check for any pending messages and process them */
 496         if (str_have_msg(&audio_str))
 497         {
 498         message_wait:
 499             /* Wait for a message to be queued */
 500             str_get_msg(&audio_str, &td.ev);
 501
 502         message_process:
 503             /* Process a message already dequeued */
 504             audio_thread_msg(&td);
 505
 506             switch (td.state)
 507             {
 508             /* These states are the only ones that should return */
 509             case TSTATE_DECODE:          goto audio_decode;
 510             case TSTATE_RENDER_WAIT:     goto render_wait;
 511             case TSTATE_RENDER_WAIT_END: goto render_wait_end;
 512             /* Anything else is interpreted as an exit */
 513             default:
 514             {
 515 #ifdef HAVE_PRIORITY_SCHEDULING
 516                 rb->thread_set_priority(THREAD_ID_CURRENT, old_priority);
 517 #endif
 518                 return;
 519                 }
 520             }
 521         }
 522
 523     audio_decode:
 524
 525         /** Buffering **/
 526         switch (audio_buffer(&audio_str, STREAM_PM_STREAMING))
 527         {
 528         case STREAM_DATA_NOT_READY:
 529         {
 530             td.state = TSTATE_DATA;
 531             goto message_wait;
 532             } /* STREAM_DATA_NOT_READY: */
 533
 534         case STREAM_DATA_END:
 535         {
 536             if (audio_queue.used > MAD_BUFFER_GUARD)
 537                 break;
 538
 539             /* Used up remainder of compressed audio buffer.
 540              * Force any residue to play if audio ended before
 541              * reaching the threshold */
 542             td.state = TSTATE_RENDER_WAIT_END;
 543             audio_queue_reset();
 544
 545         render_wait_end:
 546             pcm_output_drain();
 547
 548             while (pcm_output_used() > (ssize_t)PCMOUT_LOW_WM)
 549             {
 550                 str_get_msg_w_tmo(&audio_str, &td.ev, 1);
 551                 if (td.ev.id != SYS_TIMEOUT)
 552                     goto message_process;
 553             }
 554
 555             td.state = TSTATE_EOS;
 556             if (td.status == STREAM_PLAYING)
 557                 stream_generate_event(&audio_str, STREAM_EV_COMPLETE, 0);
 558
 559             rb->yield();
 560             goto message_wait;
 561             } /* STREAM_DATA_END: */
 562         }
 563
 564         /** Decoding **/
 565         mad_stream_buffer(&stream, audio_queue.ptr, audio_queue.used);
 566
 567         int mad_stat = mad_frame_decode(&frame, &stream);
 568
 569         ssize_t len = stream.next_frame - audio_queue.ptr;
 570
 571         if (mad_stat != 0)
 572         {
 573             DEBUGF("audio: Stream error: %s\n",
 574                    mad_stream_errorstr(&stream));
 575
 576             /* If something's goofed - try to perform resync by moving
 577              * at least one byte at a time */
 578             audio_queue_advance_pos(MAX(len, 1));
 579
 580             if (stream.error == MAD_FLAG_INCOMPLETE
 581                 || stream.error == MAD_ERROR_BUFLEN)
 582             {
 583                 /* This makes the codec support partially corrupted files */
 584                 if (++td.mad_errors <= MPA_MAX_FRAME_SIZE)
 585                 {
 586                     stream.error = 0;
 587                     rb->yield();
 588                     continue;
 589                 }
 590                 DEBUGF("audio: Too many errors\n");
 591             }
 592             else if (MAD_RECOVERABLE(stream.error))
 593             {
 594                 /* libmad says it can recover - just keep on decoding */
 595                 rb->yield();
 596                 continue;
 597             }
 598             else
 599             {
 600                 /* Some other unrecoverable error */
 601                 DEBUGF("audio: Unrecoverable error\n");
 602             }
 603
 604             /* This is too hard - bail out */
 605             td.state = TSTATE_EOS;
 606
 607             if (td.status == STREAM_PLAYING)
 608                 stream_generate_event(&audio_str, STREAM_EV_COMPLETE, 0);
 609
 610             td.status = STREAM_ERROR;
 611             goto message_wait;
 612         }
 613
 614         /* Adjust sizes by the frame size */
 615         audio_queue_advance_pos(len);
 616         td.mad_errors = 0; /* Clear errors */
 617
 618         /* Generate the pcm samples */
 619         mad_synth_frame(&synth, &frame);
 620
 621         /** Output **/
 622         if (frame.header.samplerate != td.samplerate)
 623         {
 624             td.samplerate = frame.header.samplerate;
 625             rb->dsp_configure(td.dsp, DSP_SWITCH_FREQUENCY,
 626                               td.samplerate);
 627         }
 628
 629         if (MAD_NCHANNELS(&frame.header) != td.nchannels)
 630         {
 631             td.nchannels = MAD_NCHANNELS(&frame.header);
 632             rb->dsp_configure(td.dsp, DSP_SET_STEREO_MODE,
 633                               td.nchannels == 1 ?
 634                                 STEREO_MONO : STEREO_NONINTERLEAVED);
 635         }
 636
 637         td.state  = TSTATE_RENDER_WAIT;
 638
 639         /* Add a frame of audio to the pcm buffer. Maximum is 1152 samples. */
 640     render_wait:
 641         if (synth.pcm.length > 0)
 642         {
 643             struct pcm_frame_header *dst_hdr = pcm_output_get_buffer();
 644             const char *src[2] =
 645                 { (char *)synth.pcm.samples[0], (char *)synth.pcm.samples[1] };
 646             int out_count = (synth.pcm.length * CLOCK_RATE
 647                                 + (td.samplerate - 1)) / td.samplerate;
 648             ssize_t size = sizeof(*dst_hdr) + out_count*4;
 649
 650             /* Wait for required amount of free buffer space */
 651             while (pcm_output_free() < size)
 652             {
 653                 /* Wait one frame */
 654                 int timeout = out_count*HZ / td.samplerate;
 655                 str_get_msg_w_tmo(&audio_str, &td.ev, MAX(timeout, 1));
 656                 if (td.ev.id != SYS_TIMEOUT)
 657                     goto message_process;
 658             }
 659
 660             out_count = rb->dsp_process(td.dsp, dst_hdr->data, src,
 661                                         synth.pcm.length);
 662
 663             if (out_count <= 0)
 664                 break;
 665
 666             dst_hdr->size = sizeof(*dst_hdr) + out_count*4;
 667             dst_hdr->time = audio_queue.curr->time;
 668
 669             /* As long as we're on this timestamp, the time is just
 670                incremented by the number of samples */
 671             audio_queue.curr->time += out_count;
 672
 673             /* Make this data available to DMA */
 674             pcm_output_add_data();
 675         }
 676
 677         rb->yield();
 678     } /* end decoding loop */
 679 }
 680
 681 /* Initializes the audio thread resources and starts the thread */
 682 bool audio_thread_init(void)
 683 {
 684     /* Initialise the encoded audio buffer and its descriptors */
 685     audio_queue.start = mpeg_malloc(AUDIOBUF_ALLOC_SIZE,
 686                                     MPEG_ALLOC_AUDIOBUF);
 687     if (audio_queue.start == NULL)
 688         return false;
 689
 690     /* Start the audio thread */
 691     audio_str.hdr.q = &audio_str_queue;
 692     rb->queue_init(audio_str.hdr.q, false);
 693
 694     /* We steal the codec thread for audio */
 695     rb->codec_thread_do_callback(audio_thread, &audio_str.thread);
 696
 697     rb->queue_enable_queue_send(audio_str.hdr.q, &audio_str_queue_send,
 698                                 audio_str.thread);
 699
 700     /* Wait for thread to initialize */
 701     str_send_msg(&audio_str, STREAM_NULL, 0);
 702
 703     return true;
 704 }
 705
 706 /* Stops the audio thread */
 707 void audio_thread_exit(void)
 708 {
 709     if (audio_str.thread != 0)
 710     {
 711         str_post_msg(&audio_str, STREAM_QUIT, 0);
 712         rb->codec_thread_do_callback(NULL, NULL);
 713         audio_str.thread = 0;
 714     }
 715 }