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
);