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 Truncate 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;
}
 
static int ogg_speex_write(struct ast_filestream *s, struct ast_frame *f)
{
    ast_log(LOG_WARNING, "Writing 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,
    .write = ogg_speex_write,
    .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,
    .write = ogg_speex_write,
    .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,
    .write = ogg_speex_write,
    .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
);