355 lines
9.0 KiB
C
355 lines
9.0 KiB
C
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/*
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* Asterisk -- An open source telephony toolkit.
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*
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* Copyright (C) 2011-2016, Timo Teräs
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*
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* See http://www.asterisk.org for more information about
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* the Asterisk project. Please do not directly contact
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* any of the maintainers of this project for assistance;
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* the project provides a web site, mailing lists and IRC
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* channels for your use.
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*
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* This program is free software, distributed under the terms of
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* the GNU General Public License Version 2. See the LICENSE file
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* at the top of the source tree.
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*/
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/*! \file
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*
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* \brief OGG/Speex streams.
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* \arg File name extension: spx
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* \ingroup formats
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*/
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/*** MODULEINFO
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<depend>speex</depend>
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<depend>ogg</depend>
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<support_level>extended</support_level>
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***/
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#include "asterisk.h"
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#include "asterisk/mod_format.h"
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#include "asterisk/module.h"
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#include "asterisk/format_cache.h"
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#include <speex/speex_header.h>
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#include <ogg/ogg.h>
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#define BLOCK_SIZE 4096 /* buffer size for feeding OGG routines */
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#define BUF_SIZE 200
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struct speex_desc { /* format specific parameters */
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/* structures for handling the Ogg container */
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ogg_sync_state oy;
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ogg_stream_state os;
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ogg_page og;
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ogg_packet op;
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int serialno;
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/*! \brief Indicates whether an End of Stream condition has been detected. */
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int eos;
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};
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static int read_packet(struct ast_filestream *fs)
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{
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struct speex_desc *s = (struct speex_desc *)fs->_private;
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char *buffer;
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int result;
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size_t bytes;
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while (1) {
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/* Get one packet */
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result = ogg_stream_packetout(&s->os, &s->op);
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if (result > 0) {
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if (s->op.bytes >= 5 && !memcmp(s->op.packet, "Speex", 5)) {
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s->serialno = s->os.serialno;
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}
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if (s->serialno == -1 || s->os.serialno != s->serialno) {
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continue;
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}
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return 0;
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}
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if (result < 0) {
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ast_log(LOG_WARNING,
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"Corrupt or missing data at this page position; continuing...\n");
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}
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/* No more packets left in the current page... */
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if (s->eos) {
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/* No more pages left in the stream */
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return -1;
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}
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while (!s->eos) {
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/* See if OGG has any pages in it's internal buffers */
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result = ogg_sync_pageout(&s->oy, &s->og);
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if (result > 0) {
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/* Read all streams. */
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if (ogg_page_serialno(&s->og) != s->os.serialno) {
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ogg_stream_reset_serialno(&s->os, ogg_page_serialno(&s->og));
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}
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/* Yes, OGG has more pages in it's internal buffers,
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add the page to the stream state */
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result = ogg_stream_pagein(&s->os, &s->og);
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if (result == 0) {
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/* Yes, got a new, valid page */
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if (ogg_page_eos(&s->og) &&
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ogg_page_serialno(&s->og) == s->serialno)
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s->eos = 1;
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break;
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}
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ast_log(LOG_WARNING,
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"Invalid page in the bitstream; continuing...\n");
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}
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if (result < 0) {
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ast_log(LOG_WARNING,
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"Corrupt or missing data in bitstream; continuing...\n");
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}
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/* No, we need to read more data from the file descrptor */
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/* get a buffer from OGG to read the data into */
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buffer = ogg_sync_buffer(&s->oy, BLOCK_SIZE);
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bytes = fread(buffer, 1, BLOCK_SIZE, fs->f);
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ogg_sync_wrote(&s->oy, bytes);
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if (bytes == 0) {
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s->eos = 1;
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}
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}
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}
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}
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/*!
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* \brief Create a new OGG/Speex filestream and set it up for reading.
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* \param fs File that points to on disk storage of the OGG/Speex data.
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* \return The new filestream.
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*/
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static int ogg_speex_open(struct ast_filestream *fs)
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{
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char *buffer;
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size_t bytes;
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struct speex_desc *s = (struct speex_desc *)fs->_private;
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SpeexHeader *hdr = NULL;
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int i, result, expected_rate;
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expected_rate = ast_format_get_sample_rate(fs->fmt->format);
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s->serialno = -1;
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ogg_sync_init(&s->oy);
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buffer = ogg_sync_buffer(&s->oy, BLOCK_SIZE);
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bytes = fread(buffer, 1, BLOCK_SIZE, fs->f);
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ogg_sync_wrote(&s->oy, bytes);
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result = ogg_sync_pageout(&s->oy, &s->og);
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if (result != 1) {
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if(bytes < BLOCK_SIZE) {
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ast_log(LOG_ERROR, "Run out of data...\n");
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} else {
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ast_log(LOG_ERROR, "Input does not appear to be an Ogg bitstream.\n");
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}
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ogg_sync_clear(&s->oy);
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return -1;
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}
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ogg_stream_init(&s->os, ogg_page_serialno(&s->og));
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if (ogg_stream_pagein(&s->os, &s->og) < 0) {
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ast_log(LOG_ERROR, "Error reading first page of Ogg bitstream data.\n");
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goto error;
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}
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if (read_packet(fs) < 0) {
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ast_log(LOG_ERROR, "Error reading initial header packet.\n");
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goto error;
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}
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hdr = speex_packet_to_header((char*)s->op.packet, s->op.bytes);
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if (memcmp(hdr->speex_string, "Speex ", 8)) {
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ast_log(LOG_ERROR, "OGG container does not contain Speex audio!\n");
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goto error;
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}
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if (hdr->frames_per_packet != 1) {
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ast_log(LOG_ERROR, "Only one frame-per-packet OGG/Speex files are currently supported!\n");
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goto error;
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}
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if (hdr->nb_channels != 1) {
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ast_log(LOG_ERROR, "Only monophonic OGG/Speex files are currently supported!\n");
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goto error;
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}
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if (hdr->rate != expected_rate) {
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ast_log(LOG_ERROR, "Unexpected sampling rate (%d != %d)!\n",
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hdr->rate, expected_rate);
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goto error;
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}
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/* this packet is the comment */
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if (read_packet(fs) < 0) {
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ast_log(LOG_ERROR, "Error reading comment packet.\n");
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goto error;
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}
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for (i = 0; i < hdr->extra_headers; i++) {
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if (read_packet(fs) < 0) {
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ast_log(LOG_ERROR, "Error reading extra header packet %d.\n", i+1);
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goto error;
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}
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}
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speex_header_free(hdr);
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return 0;
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error:
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if (hdr) {
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speex_header_free(hdr);
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}
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ogg_stream_clear(&s->os);
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ogg_sync_clear(&s->oy);
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return -1;
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}
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/*!
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* \brief Close a OGG/Speex filestream.
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* \param fs A OGG/Speex filestream.
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*/
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static void ogg_speex_close(struct ast_filestream *fs)
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{
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struct speex_desc *s = (struct speex_desc *)fs->_private;
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ogg_stream_clear(&s->os);
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ogg_sync_clear(&s->oy);
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}
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/*!
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* \brief Read a frame full of audio data from the filestream.
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* \param fs The filestream.
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* \param whennext Number of sample times to schedule the next call.
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* \return A pointer to a frame containing audio data or NULL ifthere is no more audio data.
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*/
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static struct ast_frame *ogg_speex_read(struct ast_filestream *fs,
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int *whennext)
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{
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struct speex_desc *s = (struct speex_desc *)fs->_private;
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if (read_packet(fs) < 0) {
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return NULL;
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}
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AST_FRAME_SET_BUFFER(&fs->fr, fs->buf, AST_FRIENDLY_OFFSET, BUF_SIZE);
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memcpy(fs->fr.data.ptr, s->op.packet, s->op.bytes);
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fs->fr.datalen = s->op.bytes;
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fs->fr.samples = *whennext = ast_codec_samples_count(&fs->fr);
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return &fs->fr;
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}
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/*!
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* \brief Truncate an OGG/Speex filestream.
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* \param s The filestream to truncate.
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* \return 0 on success, -1 on failure.
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*/
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static int ogg_speex_trunc(struct ast_filestream *s)
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{
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ast_log(LOG_WARNING, "Truncation is not supported on OGG/Speex streams!\n");
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return -1;
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}
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static int ogg_speex_write(struct ast_filestream *s, struct ast_frame *f)
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{
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ast_log(LOG_WARNING, "Writing is not supported on OGG/Speex streams!\n");
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return -1;
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}
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/*!
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* \brief Seek to a specific position in an OGG/Speex filestream.
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* \param s The filestream to truncate.
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* \param sample_offset New position for the filestream, measured in 8KHz samples.
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* \param whence Location to measure
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* \return 0 on success, -1 on failure.
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*/
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static int ogg_speex_seek(struct ast_filestream *s, off_t sample_offset, int whence)
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{
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ast_log(LOG_WARNING, "Seeking is not supported on OGG/Speex streams!\n");
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return -1;
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}
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static off_t ogg_speex_tell(struct ast_filestream *s)
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{
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ast_log(LOG_WARNING, "Telling is not supported on OGG/Speex streams!\n");
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return -1;
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}
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static struct ast_format_def speex_f = {
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.name = "ogg_speex",
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.exts = "spx",
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.open = ogg_speex_open,
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.write = ogg_speex_write,
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.seek = ogg_speex_seek,
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.trunc = ogg_speex_trunc,
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.tell = ogg_speex_tell,
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.read = ogg_speex_read,
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.close = ogg_speex_close,
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.buf_size = BUF_SIZE + AST_FRIENDLY_OFFSET,
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.desc_size = sizeof(struct speex_desc),
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};
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static struct ast_format_def speex16_f = {
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.name = "ogg_speex16",
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.exts = "spx16",
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.open = ogg_speex_open,
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.write = ogg_speex_write,
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.seek = ogg_speex_seek,
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.trunc = ogg_speex_trunc,
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.tell = ogg_speex_tell,
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.read = ogg_speex_read,
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.close = ogg_speex_close,
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.buf_size = BUF_SIZE + AST_FRIENDLY_OFFSET,
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.desc_size = sizeof(struct speex_desc),
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};
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static struct ast_format_def speex32_f = {
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.name = "ogg_speex32",
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.exts = "spx32",
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.open = ogg_speex_open,
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.write = ogg_speex_write,
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.seek = ogg_speex_seek,
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.trunc = ogg_speex_trunc,
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.tell = ogg_speex_tell,
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.read = ogg_speex_read,
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.close = ogg_speex_close,
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.buf_size = BUF_SIZE + AST_FRIENDLY_OFFSET,
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.desc_size = sizeof(struct speex_desc),
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};
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static int unload_module(void)
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{
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int res = 0;
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res |= ast_format_def_unregister(speex_f.name);
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res |= ast_format_def_unregister(speex16_f.name);
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res |= ast_format_def_unregister(speex32_f.name);
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return res;
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}
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static int load_module(void)
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{
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speex_f.format = ast_format_speex;
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speex16_f.format = ast_format_speex16;
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speex32_f.format = ast_format_speex32;
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if (ast_format_def_register(&speex_f) ||
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ast_format_def_register(&speex16_f) ||
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ast_format_def_register(&speex32_f)) {
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unload_module();
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return AST_MODULE_LOAD_DECLINE;
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}
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return AST_MODULE_LOAD_SUCCESS;
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}
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AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "OGG/Speex audio",
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.support_level = AST_MODULE_SUPPORT_EXTENDED,
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.load = load_module,
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.unload = unload_module,
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.load_pri = AST_MODPRI_APP_DEPEND
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);
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