summaryrefslogtreecommitdiffstats
path: root/testing/asterisk/ASTERISK-18995.patch
diff options
context:
space:
mode:
Diffstat (limited to 'testing/asterisk/ASTERISK-18995.patch')
-rw-r--r--testing/asterisk/ASTERISK-18995.patch358
1 files changed, 0 insertions, 358 deletions
diff --git a/testing/asterisk/ASTERISK-18995.patch b/testing/asterisk/ASTERISK-18995.patch
deleted file mode 100644
index cd144847b..000000000
--- a/testing/asterisk/ASTERISK-18995.patch
+++ /dev/null
@@ -1,358 +0,0 @@
---- /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
-+);
generated by cgit v1.2.3 (git 2.25.1) at 2024-06-28 10:52:32 +0000