format_ogg_speex.c

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/*
 * Asterisk -- An open source telephony toolkit.
 *
 * Copyright (C) 2011-2016, Timo Teräs
 *
 * See http://www.asterisk.org for more information about
 * the Asterisk project. Please do not directly contact
 * any of the maintainers of this project for assistance;
 * the project provides a web site, mailing lists and IRC
 * channels for your use.
 *
 * This program is free software, distributed under the terms of
 * the GNU General Public License Version 2. See the LICENSE file
 * at the top of the source tree.
 */

/*! \file
 *
 * \brief OGG/Speex streams.
 * \arg File name extension: spx
 * \ingroup formats
 */

/*** MODULEINFO
   <depend>speex</depend>
   <depend>ogg</depend>
   <support_level>extended</support_level>
 ***/

#include "asterisk.h"

#include "asterisk/mod_format.h"
#include "asterisk/module.h"
#include "asterisk/format_cache.h"

#include <speex/speex_header.h>
#include <ogg/ogg.h>

#define BLOCK_SIZE   4096     /* buffer size for feeding OGG routines */
#define  BUF_SIZE 200

struct speex_desc {  /* format specific parameters */
   /* structures for handling the Ogg container */
   ogg_sync_state oy;
   ogg_stream_state os;
   ogg_page og;
   ogg_packet op;

   int serialno;

   /*! \brief Indicates whether an End of Stream condition has been detected. */
   int eos;
};

static int read_packet(struct ast_filestream *fs)
{
   struct speex_desc *s = (struct speex_desc *)fs->_private;
   char *buffer;
   int result;
   size_t bytes;

   while (1) {
      /* Get one packet */
      result = ogg_stream_packetout(&s->os, &s->op);
      if (result > 0) {
         if (s->op.bytes >= 5 && !memcmp(s->op.packet, "Speex", 5)) {
            s->serialno = s->os.serialno;
         }
         if (s->serialno == -1 || s->os.serialno != s->serialno) {
            continue;
         }
         return 0;
      }

      if (result < 0) {
         ast_log(LOG_WARNING,
            "Corrupt or missing data at this page position; continuing...\n");
      }

      /* No more packets left in the current page... */
      if (s->eos) {
         /* No more pages left in the stream */
         return -1;
      }

      while (!s->eos) {
         /* See if OGG has any pages in it's internal buffers */
         result = ogg_sync_pageout(&s->oy, &s->og);
         if (result > 0) {
            /* Read all streams. */
            if (ogg_page_serialno(&s->og) != s->os.serialno) {
               ogg_stream_reset_serialno(&s->os, ogg_page_serialno(&s->og));
            }
            /* Yes, OGG has more pages in it's internal buffers,
               add the page to the stream state */
            result = ogg_stream_pagein(&s->os, &s->og);
            if (result == 0) {
               /* Yes, got a new, valid page */
               if (ogg_page_eos(&s->og) &&
                   ogg_page_serialno(&s->og) == s->serialno)
                  s->eos = 1;
               break;
            }
            ast_log(LOG_WARNING,
               "Invalid page in the bitstream; continuing...\n");
         }

         if (result < 0) {
            ast_log(LOG_WARNING,
               "Corrupt or missing data in bitstream; continuing...\n");
         }

         /* No, we need to read more data from the file descrptor */
         /* get a buffer from OGG to read the data into */
         buffer = ogg_sync_buffer(&s->oy, BLOCK_SIZE);
         bytes = fread(buffer, 1, BLOCK_SIZE, fs->f);
         ogg_sync_wrote(&s->oy, bytes);
         if (bytes == 0) {
            s->eos = 1;
         }
      }
   }
}

/*!
 * \brief Create a new OGG/Speex filestream and set it up for reading.
 * \param fs File that points to on disk storage of the OGG/Speex data.
 * \return The new filestream.
 */
static int ogg_speex_open(struct ast_filestream *fs)
{
   char *buffer;
   size_t bytes;
   struct speex_desc *s = (struct speex_desc *)fs->_private;
   SpeexHeader *hdr = NULL;
   int i, result, expected_rate;

   expected_rate = ast_format_get_sample_rate(fs->fmt->format);
   s->serialno = -1;
   ogg_sync_init(&s->oy);

   buffer = ogg_sync_buffer(&s->oy, BLOCK_SIZE);
   bytes = fread(buffer, 1, BLOCK_SIZE, fs->f);
   ogg_sync_wrote(&s->oy, bytes);

   result = ogg_sync_pageout(&s->oy, &s->og);
   if (result != 1) {
      if(bytes < BLOCK_SIZE) {
         ast_log(LOG_ERROR, "Run out of data...\n");
      } else {
         ast_log(LOG_ERROR, "Input does not appear to be an Ogg bitstream.\n");
      }
      ogg_sync_clear(&s->oy);
      return -1;
   }

   ogg_stream_init(&s->os, ogg_page_serialno(&s->og));
   if (ogg_stream_pagein(&s->os, &s->og) < 0) {
      ast_log(LOG_ERROR, "Error reading first page of Ogg bitstream data.\n");
      goto error;
   }

   if (read_packet(fs) < 0) {
      ast_log(LOG_ERROR, "Error reading initial header packet.\n");
      goto error;
   }

   hdr = speex_packet_to_header((char*)s->op.packet, s->op.bytes);
   if (memcmp(hdr->speex_string, "Speex   ", 8)) {
      ast_log(LOG_ERROR, "OGG container does not contain Speex audio!\n");
      goto error;
   }
   if (hdr->frames_per_packet != 1) {
      ast_log(LOG_ERROR, "Only one frame-per-packet OGG/Speex files are currently supported!\n");
      goto error;
   }
   if (hdr->nb_channels != 1) {
      ast_log(LOG_ERROR, "Only monophonic OGG/Speex files are currently supported!\n");
      goto error;
   }
   if (hdr->rate != expected_rate) {
      ast_log(LOG_ERROR, "Unexpected sampling rate (%d != %d)!\n",
         hdr->rate, expected_rate);
      goto error;
   }

   /* this packet is the comment */
   if (read_packet(fs) < 0) {
      ast_log(LOG_ERROR, "Error reading comment packet.\n");
      goto error;
   }
   for (i = 0; i < hdr->extra_headers; i++) {
      if (read_packet(fs) < 0) {
         ast_log(LOG_ERROR, "Error reading extra header packet %d.\n", i+1);
         goto error;
      }
   }
   speex_header_free(hdr);

   return 0;
error:
   if (hdr) {
      speex_header_free(hdr);
   }
   ogg_stream_clear(&s->os);
   ogg_sync_clear(&s->oy);
   return -1;
}

/*!
 * \brief Close a OGG/Speex filestream.
 * \param fs A OGG/Speex filestream.
 */
static void ogg_speex_close(struct ast_filestream *fs)
{
   struct speex_desc *s = (struct speex_desc *)fs->_private;

   ogg_stream_clear(&s->os);
   ogg_sync_clear(&s->oy);
}

/*!
 * \brief Read a frame full of audio data from the filestream.
 * \param fs The filestream.
 * \param whennext Number of sample times to schedule the next call.
 * \return A pointer to a frame containing audio data or NULL ifthere is no more audio data.
 */
static struct ast_frame *ogg_speex_read(struct ast_filestream *fs,
                int *whennext)
{
   struct speex_desc *s = (struct speex_desc *)fs->_private;

   if (read_packet(fs) < 0) {
      return NULL;
   }

   AST_FRAME_SET_BUFFER(&fs->fr, fs->buf, AST_FRIENDLY_OFFSET, BUF_SIZE);
   memcpy(fs->fr.data.ptr, s->op.packet, s->op.bytes);
   fs->fr.datalen = s->op.bytes;
   fs->fr.samples = *whennext = ast_codec_samples_count(&fs->fr);

   return &fs->fr;
}

/*!
 * \brief Trucate an OGG/Speex filestream.
 * \param s The filestream to truncate.
 * \return 0 on success, -1 on failure.
 */

static int ogg_speex_trunc(struct ast_filestream *s)
{
   ast_log(LOG_WARNING, "Truncation is not supported on OGG/Speex streams!\n");
   return -1;
}

/*!
 * \brief Seek to a specific position in an OGG/Speex filestream.
 * \param s The filestream to truncate.
 * \param sample_offset New position for the filestream, measured in 8KHz samples.
 * \param whence Location to measure
 * \return 0 on success, -1 on failure.
 */
static int ogg_speex_seek(struct ast_filestream *s, off_t sample_offset, int whence)
{
   ast_log(LOG_WARNING, "Seeking is not supported on OGG/Speex streams!\n");
   return -1;
}

static off_t ogg_speex_tell(struct ast_filestream *s)
{
   ast_log(LOG_WARNING, "Telling is not supported on OGG/Speex streams!\n");
   return -1;
}

static struct ast_format_def speex_f = {
   .name = "ogg_speex",
   .exts = "spx",
   .open = ogg_speex_open,
   .seek = ogg_speex_seek,
   .trunc = ogg_speex_trunc,
   .tell = ogg_speex_tell,
   .read = ogg_speex_read,
   .close = ogg_speex_close,
   .buf_size = BUF_SIZE + AST_FRIENDLY_OFFSET,
   .desc_size = sizeof(struct speex_desc),
};

static struct ast_format_def speex16_f = {
   .name = "ogg_speex16",
   .exts = "spx16",
   .open = ogg_speex_open,
   .seek = ogg_speex_seek,
   .trunc = ogg_speex_trunc,
   .tell = ogg_speex_tell,
   .read = ogg_speex_read,
   .close = ogg_speex_close,
   .buf_size = BUF_SIZE + AST_FRIENDLY_OFFSET,
   .desc_size = sizeof(struct speex_desc),
};

static struct ast_format_def speex32_f = {
   .name = "ogg_speex32",
   .exts = "spx32",
   .open = ogg_speex_open,
   .seek = ogg_speex_seek,
   .trunc = ogg_speex_trunc,
   .tell = ogg_speex_tell,
   .read = ogg_speex_read,
   .close = ogg_speex_close,
   .buf_size = BUF_SIZE + AST_FRIENDLY_OFFSET,
   .desc_size = sizeof(struct speex_desc),
};

static int unload_module(void)
{
   int res = 0;
   res |= ast_format_def_unregister(speex_f.name);
   res |= ast_format_def_unregister(speex16_f.name);
   res |= ast_format_def_unregister(speex32_f.name);
   return res;
}

static int load_module(void)
{
   speex_f.format = ast_format_speex;
   speex16_f.format = ast_format_speex16;
   speex32_f.format = ast_format_speex32;

   if (ast_format_def_register(&speex_f) ||
       ast_format_def_register(&speex16_f) ||
       ast_format_def_register(&speex32_f)) {
      unload_module();
      return AST_MODULE_LOAD_DECLINE;
   }

   return AST_MODULE_LOAD_SUCCESS;
}

AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "OGG/Speex audio",
   .support_level = AST_MODULE_SUPPORT_EXTENDED,
   .load = load_module,
   .unload = unload_module,
   .load_pri = AST_MODPRI_APP_DEPEND
);