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Voice
asterisk
Commits
827c47b7
Commit
827c47b7
authored
8 years ago
by
Joshua Colp
Committed by
Gerrit Code Review
8 years ago
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Merge "Add support for OGG/Speex file format"
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947f76a9
56bdf048
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formats/format_ogg_speex.c
+345
-0
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formats/format_ogg_speex.c
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827c47b7
...
@@ -249,6 +249,13 @@ Functions
...
@@ -249,6 +249,13 @@ Functions
* The func_odbc global option "single_db_connection" default value has been
* The func_odbc global option "single_db_connection" default value has been
changed to 'no'.
changed to 'no'.
Formats
------------------
* New module format_ogg_speex added which supports Speex codec inside
Ogg containers (filename extension .spx).
CHANNEL
CHANNEL
------------------
------------------
* Added CHANNEL(onhold) item that returns 1 (onhold) and 0 (not-onhold) for
* Added CHANNEL(onhold) item that returns 1 (onhold) and 0 (not-onhold) for
...
...
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formats/format_ogg_speex.c
0 → 100644
+
345
−
0
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827c47b7
/*
* 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"
ASTERISK_REGISTER_FILE
()
#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
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
))
{
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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