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-rw-r--r--main/asterisk/ASTERISK-18995.patch358
1 files changed, 358 insertions, 0 deletions
diff --git a/main/asterisk/ASTERISK-18995.patch b/main/asterisk/ASTERISK-18995.patch
new file mode 100644
index 0000000000..cd144847be
--- /dev/null
+++ b/main/asterisk/ASTERISK-18995.patch
@@ -0,0 +1,358 @@
+--- /dev/null 2011-11-29 09:02:40.279581283 +0200
++++ b/formats/format_ogg_speex.c 2011-12-08 15:57:12.000000000 +0200
+@@ -0,0 +1,355 @@
++/*
++ * Asterisk -- An open source telephony toolkit.
++ *
++ * Copyright (C) 2011, 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"
++
++ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
++
++#include "asterisk/mod_format.h"
++#include "asterisk/module.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 format_id;
++ 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.
++ * \param expected_rate The expected Speex format (sampling rate).
++ * \return The new filestream.
++ */
++static int ogg_speex_open(struct ast_filestream *fs, int format_id, int expected_rate)
++{
++ char *buffer;
++ size_t bytes;
++ struct speex_desc *s = (struct speex_desc *)fs->_private;
++ SpeexHeader *hdr = NULL;
++ int i, result;
++
++ s->format_id = format_id;
++ 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;
++ }
++ }
++ free(hdr);
++
++ return 0;
++error:
++ if (hdr)
++ 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;
++
++ fs->fr.frametype = AST_FRAME_VOICE;
++ ast_format_set(&fs->fr.subclass.format, s->format_id, 0);
++ fs->fr.mallocd = 0;
++ 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_get_samples(&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 int ogg_speex_open_nb(struct ast_filestream *fs)
++{
++ return ogg_speex_open(fs, AST_FORMAT_SPEEX, 8000);
++}
++
++static struct ast_format_def speex_f = {
++ .name = "ogg_speex",
++ .exts = "spx",
++ .open = ogg_speex_open_nb,
++ .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 ogg_speex_open_wb(struct ast_filestream *fs)
++{
++ return ogg_speex_open(fs, AST_FORMAT_SPEEX16, 16000);
++}
++
++static struct ast_format_def speex16_f = {
++ .name = "ogg_speex16",
++ .exts = "spx16",
++ .open = ogg_speex_open_wb,
++ .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 ogg_speex_open_uwb(struct ast_filestream *fs)
++{
++ return ogg_speex_open(fs, AST_FORMAT_SPEEX32, 32000);
++}
++
++static struct ast_format_def speex32_f = {
++ .name = "ogg_speex32",
++ .exts = "spx32",
++ .open = ogg_speex_open_uwb,
++ .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 load_module(void)
++{
++ ast_format_set(&speex_f.format, AST_FORMAT_SPEEX, 0);
++ ast_format_set(&speex16_f.format, AST_FORMAT_SPEEX16, 0);
++ ast_format_set(&speex32_f.format, AST_FORMAT_SPEEX32, 0);
++
++ if (ast_format_def_register(&speex_f) ||
++ ast_format_def_register(&speex16_f) ||
++ ast_format_def_register(&speex32_f))
++ return AST_MODULE_LOAD_FAILURE;
++
++ return AST_MODULE_LOAD_SUCCESS;
++}
++
++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;
++}
++
++AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "OGG/Speex audio",
++ .load = load_module,
++ .unload = unload_module,
++ .load_pri = AST_MODPRI_APP_DEPEND
++);
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