2007-08-08 19:30:52 +00:00
/*
* Asterisk - - An open source telephony toolkit .
*
* Copyright ( C ) 1999 - 2007 , Digium , Inc .
*
* Joshua Colp < jcolp @ digium . com >
*
* 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 Audiohooks Architecture
*
2009-05-01 14:58:59 +00:00
* \ author Joshua Colp < jcolp @ digium . com >
2007-08-08 19:30:52 +00:00
*/
# include "asterisk.h"
ASTERISK_FILE_VERSION ( __FILE__ , " $Revision$ " )
# include <signal.h>
# include "asterisk/channel.h"
# include "asterisk/utils.h"
# include "asterisk/lock.h"
# include "asterisk/linkedlists.h"
# include "asterisk/audiohook.h"
# include "asterisk/slinfactory.h"
# include "asterisk/frame.h"
# include "asterisk/translate.h"
Media Project Phase2: SILK 8khz-24khz, SLINEAR 8khz-192khz, SPEEX 32khz, hd audio ConfBridge, and other stuff
-Functional changes
1. Dynamic global format list build by codecs defined in codecs.conf
2. SILK 8khz, 12khz, 16khz, and 24khz with custom attributes defined in codecs.conf
3. Negotiation of SILK attributes in chan_sip.
4. SPEEX 32khz with translation
5. SLINEAR 8khz, 12khz, 24khz, 32khz, 44.1khz, 48khz, 96khz, 192khz with translation
using codec_resample.c
6. Various changes to RTP code required to properly handle the dynamic format list
and formats with attributes.
7. ConfBridge now dynamically jumps to the best possible sample rate. This allows
for conferences to take advantage of HD audio (Which sounds awesome)
8. Audiohooks are no longer limited to 8khz audio, and most effects have been
updated to take advantage of this such as Volume, DENOISE, PITCH_SHIFT.
9. codec_resample now uses its own code rather than depending on libresample.
-Organizational changes
Global format list is moved from frame.c to format.c
Various format specific functions moved from frame.c to format.c
Review: https://reviewboard.asterisk.org/r/1104/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308582 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-22 23:04:49 +00:00
# define AST_AUDIOHOOK_SYNC_TOLERANCE 100 /*!< Tolerance in milliseconds for audiohooks synchronization */
# define AST_AUDIOHOOK_SMALL_QUEUE_TOLERANCE 100 /*!< When small queue is enabled, this is the maximum amount of audio that can remain queued at a time. */
2007-08-08 19:30:52 +00:00
struct ast_audiohook_translate {
struct ast_trans_pvt * trans_pvt ;
2011-02-03 16:22:10 +00:00
struct ast_format format ;
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} ;
struct ast_audiohook_list {
Media Project Phase2: SILK 8khz-24khz, SLINEAR 8khz-192khz, SPEEX 32khz, hd audio ConfBridge, and other stuff
-Functional changes
1. Dynamic global format list build by codecs defined in codecs.conf
2. SILK 8khz, 12khz, 16khz, and 24khz with custom attributes defined in codecs.conf
3. Negotiation of SILK attributes in chan_sip.
4. SPEEX 32khz with translation
5. SLINEAR 8khz, 12khz, 24khz, 32khz, 44.1khz, 48khz, 96khz, 192khz with translation
using codec_resample.c
6. Various changes to RTP code required to properly handle the dynamic format list
and formats with attributes.
7. ConfBridge now dynamically jumps to the best possible sample rate. This allows
for conferences to take advantage of HD audio (Which sounds awesome)
8. Audiohooks are no longer limited to 8khz audio, and most effects have been
updated to take advantage of this such as Volume, DENOISE, PITCH_SHIFT.
9. codec_resample now uses its own code rather than depending on libresample.
-Organizational changes
Global format list is moved from frame.c to format.c
Various format specific functions moved from frame.c to format.c
Review: https://reviewboard.asterisk.org/r/1104/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308582 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-22 23:04:49 +00:00
/* If all the audiohooks in this list are capable
* of processing slinear at any sample rate , this
* variable will be set and the sample rate will
* be preserved during ast_audiohook_write_list ( ) */
int native_slin_compatible ;
int list_internal_samp_rate ; /*!< Internal sample rate used when writing to the audiohook list */
2007-08-08 19:30:52 +00:00
struct ast_audiohook_translate in_translate [ 2 ] ;
struct ast_audiohook_translate out_translate [ 2 ] ;
AST_LIST_HEAD_NOLOCK ( , ast_audiohook ) spy_list ;
AST_LIST_HEAD_NOLOCK ( , ast_audiohook ) whisper_list ;
AST_LIST_HEAD_NOLOCK ( , ast_audiohook ) manipulate_list ;
} ;
Media Project Phase2: SILK 8khz-24khz, SLINEAR 8khz-192khz, SPEEX 32khz, hd audio ConfBridge, and other stuff
-Functional changes
1. Dynamic global format list build by codecs defined in codecs.conf
2. SILK 8khz, 12khz, 16khz, and 24khz with custom attributes defined in codecs.conf
3. Negotiation of SILK attributes in chan_sip.
4. SPEEX 32khz with translation
5. SLINEAR 8khz, 12khz, 24khz, 32khz, 44.1khz, 48khz, 96khz, 192khz with translation
using codec_resample.c
6. Various changes to RTP code required to properly handle the dynamic format list
and formats with attributes.
7. ConfBridge now dynamically jumps to the best possible sample rate. This allows
for conferences to take advantage of HD audio (Which sounds awesome)
8. Audiohooks are no longer limited to 8khz audio, and most effects have been
updated to take advantage of this such as Volume, DENOISE, PITCH_SHIFT.
9. codec_resample now uses its own code rather than depending on libresample.
-Organizational changes
Global format list is moved from frame.c to format.c
Various format specific functions moved from frame.c to format.c
Review: https://reviewboard.asterisk.org/r/1104/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308582 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-22 23:04:49 +00:00
static int audiohook_set_internal_rate ( struct ast_audiohook * audiohook , int rate , int reset )
{
struct ast_format slin ;
if ( audiohook - > hook_internal_samp_rate = = rate ) {
return 0 ;
}
audiohook - > hook_internal_samp_rate = rate ;
ast_format_set ( & slin , ast_format_slin_by_rate ( rate ) , 0 ) ;
/* Setup the factories that are needed for this audiohook type */
switch ( audiohook - > type ) {
case AST_AUDIOHOOK_TYPE_SPY :
if ( reset ) {
ast_slinfactory_destroy ( & audiohook - > read_factory ) ;
}
ast_slinfactory_init_with_format ( & audiohook - > read_factory , & slin ) ;
/* fall through */
case AST_AUDIOHOOK_TYPE_WHISPER :
if ( reset ) {
ast_slinfactory_destroy ( & audiohook - > write_factory ) ;
}
ast_slinfactory_init_with_format ( & audiohook - > write_factory , & slin ) ;
break ;
default :
break ;
}
return 0 ;
}
2007-08-08 19:30:52 +00:00
/*! \brief Initialize an audiohook structure
* \ param audiohook Audiohook structure
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* \ param type
* \ param source
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* \ return Returns 0 on success , - 1 on failure
*/
Media Project Phase2: SILK 8khz-24khz, SLINEAR 8khz-192khz, SPEEX 32khz, hd audio ConfBridge, and other stuff
-Functional changes
1. Dynamic global format list build by codecs defined in codecs.conf
2. SILK 8khz, 12khz, 16khz, and 24khz with custom attributes defined in codecs.conf
3. Negotiation of SILK attributes in chan_sip.
4. SPEEX 32khz with translation
5. SLINEAR 8khz, 12khz, 24khz, 32khz, 44.1khz, 48khz, 96khz, 192khz with translation
using codec_resample.c
6. Various changes to RTP code required to properly handle the dynamic format list
and formats with attributes.
7. ConfBridge now dynamically jumps to the best possible sample rate. This allows
for conferences to take advantage of HD audio (Which sounds awesome)
8. Audiohooks are no longer limited to 8khz audio, and most effects have been
updated to take advantage of this such as Volume, DENOISE, PITCH_SHIFT.
9. codec_resample now uses its own code rather than depending on libresample.
-Organizational changes
Global format list is moved from frame.c to format.c
Various format specific functions moved from frame.c to format.c
Review: https://reviewboard.asterisk.org/r/1104/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308582 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-22 23:04:49 +00:00
int ast_audiohook_init ( struct ast_audiohook * audiohook , enum ast_audiohook_type type , const char * source , enum ast_audiohook_init_flags init_flags )
2007-08-08 19:30:52 +00:00
{
/* Need to keep the type and source */
audiohook - > type = type ;
audiohook - > source = source ;
/* Initialize lock that protects our audiohook */
ast_mutex_init ( & audiohook - > lock ) ;
ast_cond_init ( & audiohook - > trigger , NULL ) ;
Media Project Phase2: SILK 8khz-24khz, SLINEAR 8khz-192khz, SPEEX 32khz, hd audio ConfBridge, and other stuff
-Functional changes
1. Dynamic global format list build by codecs defined in codecs.conf
2. SILK 8khz, 12khz, 16khz, and 24khz with custom attributes defined in codecs.conf
3. Negotiation of SILK attributes in chan_sip.
4. SPEEX 32khz with translation
5. SLINEAR 8khz, 12khz, 24khz, 32khz, 44.1khz, 48khz, 96khz, 192khz with translation
using codec_resample.c
6. Various changes to RTP code required to properly handle the dynamic format list
and formats with attributes.
7. ConfBridge now dynamically jumps to the best possible sample rate. This allows
for conferences to take advantage of HD audio (Which sounds awesome)
8. Audiohooks are no longer limited to 8khz audio, and most effects have been
updated to take advantage of this such as Volume, DENOISE, PITCH_SHIFT.
9. codec_resample now uses its own code rather than depending on libresample.
-Organizational changes
Global format list is moved from frame.c to format.c
Various format specific functions moved from frame.c to format.c
Review: https://reviewboard.asterisk.org/r/1104/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308582 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-22 23:04:49 +00:00
audiohook - > init_flags = init_flags ;
/* initialize internal rate at 8khz, this will adjust if necessary */
audiohook_set_internal_rate ( audiohook , 8000 , 0 ) ;
2007-08-08 19:30:52 +00:00
/* Since we are just starting out... this audiohook is new */
2009-11-20 17:26:20 +00:00
ast_audiohook_update_status ( audiohook , AST_AUDIOHOOK_STATUS_NEW ) ;
2007-08-08 19:30:52 +00:00
return 0 ;
}
/*! \brief Destroys an audiohook structure
* \ param audiohook Audiohook structure
* \ return Returns 0 on success , - 1 on failure
*/
int ast_audiohook_destroy ( struct ast_audiohook * audiohook )
{
/* Drop the factories used by this audiohook type */
switch ( audiohook - > type ) {
case AST_AUDIOHOOK_TYPE_SPY :
ast_slinfactory_destroy ( & audiohook - > read_factory ) ;
case AST_AUDIOHOOK_TYPE_WHISPER :
ast_slinfactory_destroy ( & audiohook - > write_factory ) ;
break ;
default :
break ;
}
/* Destroy translation path if present */
if ( audiohook - > trans_pvt )
ast_translator_free_path ( audiohook - > trans_pvt ) ;
/* Lock and trigger be gone! */
ast_cond_destroy ( & audiohook - > trigger ) ;
ast_mutex_destroy ( & audiohook - > lock ) ;
return 0 ;
}
/*! \brief Writes a frame into the audiohook structure
* \ param audiohook Audiohook structure
* \ param direction Direction the audio frame came from
* \ param frame Frame to write in
* \ return Returns 0 on success , - 1 on failure
*/
int ast_audiohook_write_frame ( struct ast_audiohook * audiohook , enum ast_audiohook_direction direction , struct ast_frame * frame )
{
struct ast_slinfactory * factory = ( direction = = AST_AUDIOHOOK_DIRECTION_READ ? & audiohook - > read_factory : & audiohook - > write_factory ) ;
2008-04-08 15:05:35 +00:00
struct ast_slinfactory * other_factory = ( direction = = AST_AUDIOHOOK_DIRECTION_READ ? & audiohook - > write_factory : & audiohook - > read_factory ) ;
2008-08-10 19:35:50 +00:00
struct timeval * rwtime = ( direction = = AST_AUDIOHOOK_DIRECTION_READ ? & audiohook - > read_time : & audiohook - > write_time ) , previous_time = * rwtime ;
2009-05-28 14:58:06 +00:00
int our_factory_samples ;
2008-10-14 23:04:44 +00:00
int our_factory_ms ;
int other_factory_samples ;
int other_factory_ms ;
2010-04-21 11:27:27 +00:00
int muteme = 0 ;
2008-04-08 15:05:35 +00:00
/* Update last feeding time to be current */
2008-08-10 19:35:50 +00:00
* rwtime = ast_tvnow ( ) ;
2008-04-08 15:05:35 +00:00
2009-05-28 14:58:06 +00:00
our_factory_samples = ast_slinfactory_available ( factory ) ;
Media Project Phase2: SILK 8khz-24khz, SLINEAR 8khz-192khz, SPEEX 32khz, hd audio ConfBridge, and other stuff
-Functional changes
1. Dynamic global format list build by codecs defined in codecs.conf
2. SILK 8khz, 12khz, 16khz, and 24khz with custom attributes defined in codecs.conf
3. Negotiation of SILK attributes in chan_sip.
4. SPEEX 32khz with translation
5. SLINEAR 8khz, 12khz, 24khz, 32khz, 44.1khz, 48khz, 96khz, 192khz with translation
using codec_resample.c
6. Various changes to RTP code required to properly handle the dynamic format list
and formats with attributes.
7. ConfBridge now dynamically jumps to the best possible sample rate. This allows
for conferences to take advantage of HD audio (Which sounds awesome)
8. Audiohooks are no longer limited to 8khz audio, and most effects have been
updated to take advantage of this such as Volume, DENOISE, PITCH_SHIFT.
9. codec_resample now uses its own code rather than depending on libresample.
-Organizational changes
Global format list is moved from frame.c to format.c
Various format specific functions moved from frame.c to format.c
Review: https://reviewboard.asterisk.org/r/1104/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308582 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-22 23:04:49 +00:00
our_factory_ms = ast_tvdiff_ms ( * rwtime , previous_time ) + ( our_factory_samples / ( audiohook - > hook_internal_samp_rate / 1000 ) ) ;
2008-10-14 23:04:44 +00:00
other_factory_samples = ast_slinfactory_available ( other_factory ) ;
Media Project Phase2: SILK 8khz-24khz, SLINEAR 8khz-192khz, SPEEX 32khz, hd audio ConfBridge, and other stuff
-Functional changes
1. Dynamic global format list build by codecs defined in codecs.conf
2. SILK 8khz, 12khz, 16khz, and 24khz with custom attributes defined in codecs.conf
3. Negotiation of SILK attributes in chan_sip.
4. SPEEX 32khz with translation
5. SLINEAR 8khz, 12khz, 24khz, 32khz, 44.1khz, 48khz, 96khz, 192khz with translation
using codec_resample.c
6. Various changes to RTP code required to properly handle the dynamic format list
and formats with attributes.
7. ConfBridge now dynamically jumps to the best possible sample rate. This allows
for conferences to take advantage of HD audio (Which sounds awesome)
8. Audiohooks are no longer limited to 8khz audio, and most effects have been
updated to take advantage of this such as Volume, DENOISE, PITCH_SHIFT.
9. codec_resample now uses its own code rather than depending on libresample.
-Organizational changes
Global format list is moved from frame.c to format.c
Various format specific functions moved from frame.c to format.c
Review: https://reviewboard.asterisk.org/r/1104/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308582 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-22 23:04:49 +00:00
other_factory_ms = other_factory_samples / ( audiohook - > hook_internal_samp_rate / 1000 ) ;
2008-10-14 23:04:44 +00:00
if ( ast_test_flag ( audiohook , AST_AUDIOHOOK_TRIGGER_SYNC ) & & other_factory_samples & & ( our_factory_ms - other_factory_ms > AST_AUDIOHOOK_SYNC_TOLERANCE ) ) {
2011-02-04 16:55:39 +00:00
ast_debug ( 1 , " Flushing audiohook %p so it remains in sync \n " , audiohook ) ;
2008-04-08 15:05:35 +00:00
ast_slinfactory_flush ( factory ) ;
ast_slinfactory_flush ( other_factory ) ;
}
2007-08-08 19:30:52 +00:00
Media Project Phase2: SILK 8khz-24khz, SLINEAR 8khz-192khz, SPEEX 32khz, hd audio ConfBridge, and other stuff
-Functional changes
1. Dynamic global format list build by codecs defined in codecs.conf
2. SILK 8khz, 12khz, 16khz, and 24khz with custom attributes defined in codecs.conf
3. Negotiation of SILK attributes in chan_sip.
4. SPEEX 32khz with translation
5. SLINEAR 8khz, 12khz, 24khz, 32khz, 44.1khz, 48khz, 96khz, 192khz with translation
using codec_resample.c
6. Various changes to RTP code required to properly handle the dynamic format list
and formats with attributes.
7. ConfBridge now dynamically jumps to the best possible sample rate. This allows
for conferences to take advantage of HD audio (Which sounds awesome)
8. Audiohooks are no longer limited to 8khz audio, and most effects have been
updated to take advantage of this such as Volume, DENOISE, PITCH_SHIFT.
9. codec_resample now uses its own code rather than depending on libresample.
-Organizational changes
Global format list is moved from frame.c to format.c
Various format specific functions moved from frame.c to format.c
Review: https://reviewboard.asterisk.org/r/1104/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308582 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-22 23:04:49 +00:00
if ( ast_test_flag ( audiohook , AST_AUDIOHOOK_SMALL_QUEUE ) & & ( ( our_factory_ms > AST_AUDIOHOOK_SMALL_QUEUE_TOLERANCE ) | | ( other_factory_ms > AST_AUDIOHOOK_SMALL_QUEUE_TOLERANCE ) ) ) {
2011-02-04 16:55:39 +00:00
ast_debug ( 1 , " Audiohook %p has stale audio in its factories. Flushing them both \n " , audiohook ) ;
2009-05-28 14:58:06 +00:00
ast_slinfactory_flush ( factory ) ;
ast_slinfactory_flush ( other_factory ) ;
}
2010-04-21 11:27:27 +00:00
/* swap frame data for zeros if mute is required */
if ( ( ast_test_flag ( audiohook , AST_AUDIOHOOK_MUTE_READ ) & & ( direction = = AST_AUDIOHOOK_DIRECTION_READ ) ) | |
( ast_test_flag ( audiohook , AST_AUDIOHOOK_MUTE_WRITE ) & & ( direction = = AST_AUDIOHOOK_DIRECTION_WRITE ) ) | |
( ast_test_flag ( audiohook , AST_AUDIOHOOK_MUTE_READ | AST_AUDIOHOOK_MUTE_WRITE ) = = ( AST_AUDIOHOOK_MUTE_READ | AST_AUDIOHOOK_MUTE_WRITE ) ) ) {
muteme = 1 ;
}
if ( muteme & & frame - > datalen > 0 ) {
ast_frame_clear ( frame ) ;
}
2007-08-08 19:30:52 +00:00
/* Write frame out to respective factory */
ast_slinfactory_feed ( factory , frame ) ;
/* If we need to notify the respective handler of this audiohook, do so */
2008-03-12 18:29:33 +00:00
if ( ( ast_test_flag ( audiohook , AST_AUDIOHOOK_TRIGGER_MODE ) = = AST_AUDIOHOOK_TRIGGER_READ ) & & ( direction = = AST_AUDIOHOOK_DIRECTION_READ ) ) {
ast_cond_signal ( & audiohook - > trigger ) ;
} else if ( ( ast_test_flag ( audiohook , AST_AUDIOHOOK_TRIGGER_MODE ) = = AST_AUDIOHOOK_TRIGGER_WRITE ) & & ( direction = = AST_AUDIOHOOK_DIRECTION_WRITE ) ) {
ast_cond_signal ( & audiohook - > trigger ) ;
} else if ( ast_test_flag ( audiohook , AST_AUDIOHOOK_TRIGGER_SYNC ) ) {
ast_cond_signal ( & audiohook - > trigger ) ;
2007-08-08 19:30:52 +00:00
}
return 0 ;
}
static struct ast_frame * audiohook_read_frame_single ( struct ast_audiohook * audiohook , size_t samples , enum ast_audiohook_direction direction )
{
struct ast_slinfactory * factory = ( direction = = AST_AUDIOHOOK_DIRECTION_READ ? & audiohook - > read_factory : & audiohook - > write_factory ) ;
int vol = ( direction = = AST_AUDIOHOOK_DIRECTION_READ ? audiohook - > options . read_volume : audiohook - > options . write_volume ) ;
short buf [ samples ] ;
struct ast_frame frame = {
. frametype = AST_FRAME_VOICE ,
2008-05-22 16:29:54 +00:00
. data . ptr = buf ,
2007-08-08 19:30:52 +00:00
. datalen = sizeof ( buf ) ,
. samples = samples ,
} ;
Media Project Phase2: SILK 8khz-24khz, SLINEAR 8khz-192khz, SPEEX 32khz, hd audio ConfBridge, and other stuff
-Functional changes
1. Dynamic global format list build by codecs defined in codecs.conf
2. SILK 8khz, 12khz, 16khz, and 24khz with custom attributes defined in codecs.conf
3. Negotiation of SILK attributes in chan_sip.
4. SPEEX 32khz with translation
5. SLINEAR 8khz, 12khz, 24khz, 32khz, 44.1khz, 48khz, 96khz, 192khz with translation
using codec_resample.c
6. Various changes to RTP code required to properly handle the dynamic format list
and formats with attributes.
7. ConfBridge now dynamically jumps to the best possible sample rate. This allows
for conferences to take advantage of HD audio (Which sounds awesome)
8. Audiohooks are no longer limited to 8khz audio, and most effects have been
updated to take advantage of this such as Volume, DENOISE, PITCH_SHIFT.
9. codec_resample now uses its own code rather than depending on libresample.
-Organizational changes
Global format list is moved from frame.c to format.c
Various format specific functions moved from frame.c to format.c
Review: https://reviewboard.asterisk.org/r/1104/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308582 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-22 23:04:49 +00:00
ast_format_set ( & frame . subclass . format , ast_format_slin_by_rate ( audiohook - > hook_internal_samp_rate ) , 0 ) ;
2007-08-08 19:30:52 +00:00
/* Ensure the factory is able to give us the samples we want */
if ( samples > ast_slinfactory_available ( factory ) )
return NULL ;
/* Read data in from factory */
if ( ! ast_slinfactory_read ( factory , buf , samples ) )
return NULL ;
/* If a volume adjustment needs to be applied apply it */
if ( vol )
ast_frame_adjust_volume ( & frame , vol ) ;
return ast_frdup ( & frame ) ;
}
2011-03-11 18:54:45 +00:00
static struct ast_frame * audiohook_read_frame_both ( struct ast_audiohook * audiohook , size_t samples , struct ast_frame * * read_reference , struct ast_frame * * write_reference )
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{
2008-03-12 18:29:33 +00:00
int i = 0 , usable_read , usable_write ;
2007-08-08 19:30:52 +00:00
short buf1 [ samples ] , buf2 [ samples ] , * read_buf = NULL , * write_buf = NULL , * final_buf = NULL , * data1 = NULL , * data2 = NULL ;
struct ast_frame frame = {
. frametype = AST_FRAME_VOICE ,
2008-05-22 16:29:54 +00:00
. data . ptr = NULL ,
2007-08-08 19:30:52 +00:00
. datalen = sizeof ( buf1 ) ,
. samples = samples ,
} ;
Media Project Phase2: SILK 8khz-24khz, SLINEAR 8khz-192khz, SPEEX 32khz, hd audio ConfBridge, and other stuff
-Functional changes
1. Dynamic global format list build by codecs defined in codecs.conf
2. SILK 8khz, 12khz, 16khz, and 24khz with custom attributes defined in codecs.conf
3. Negotiation of SILK attributes in chan_sip.
4. SPEEX 32khz with translation
5. SLINEAR 8khz, 12khz, 24khz, 32khz, 44.1khz, 48khz, 96khz, 192khz with translation
using codec_resample.c
6. Various changes to RTP code required to properly handle the dynamic format list
and formats with attributes.
7. ConfBridge now dynamically jumps to the best possible sample rate. This allows
for conferences to take advantage of HD audio (Which sounds awesome)
8. Audiohooks are no longer limited to 8khz audio, and most effects have been
updated to take advantage of this such as Volume, DENOISE, PITCH_SHIFT.
9. codec_resample now uses its own code rather than depending on libresample.
-Organizational changes
Global format list is moved from frame.c to format.c
Various format specific functions moved from frame.c to format.c
Review: https://reviewboard.asterisk.org/r/1104/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308582 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-22 23:04:49 +00:00
ast_format_set ( & frame . subclass . format , ast_format_slin_by_rate ( audiohook - > hook_internal_samp_rate ) , 0 ) ;
2007-08-08 19:30:52 +00:00
2008-03-12 18:29:33 +00:00
/* Make sure both factories have the required samples */
usable_read = ( ast_slinfactory_available ( & audiohook - > read_factory ) > = samples ? 1 : 0 ) ;
usable_write = ( ast_slinfactory_available ( & audiohook - > write_factory ) > = samples ? 1 : 0 ) ;
if ( ! usable_read & & ! usable_write ) {
/* If both factories are unusable bail out */
ast_debug ( 1 , " Read factory %p and write factory %p both fail to provide %zd samples \n " , & audiohook - > read_factory , & audiohook - > write_factory , samples ) ;
return NULL ;
}
/* If we want to provide only a read factory make sure we aren't waiting for other audio */
if ( usable_read & & ! usable_write & & ( ast_tvdiff_ms ( ast_tvnow ( ) , audiohook - > write_time ) < ( samples / 8 ) * 2 ) ) {
2008-07-14 10:39:23 +00:00
ast_debug ( 3 , " Write factory %p was pretty quick last time, waiting for them. \n " , & audiohook - > write_factory ) ;
2008-03-12 18:29:33 +00:00
return NULL ;
}
/* If we want to provide only a write factory make sure we aren't waiting for other audio */
2008-07-11 19:14:15 +00:00
if ( usable_write & & ! usable_read & & ( ast_tvdiff_ms ( ast_tvnow ( ) , audiohook - > read_time ) < ( samples / 8 ) * 2 ) ) {
2008-07-14 10:39:23 +00:00
ast_debug ( 3 , " Read factory %p was pretty quick last time, waiting for them. \n " , & audiohook - > read_factory ) ;
2008-03-12 18:29:33 +00:00
return NULL ;
}
2007-08-08 19:30:52 +00:00
/* Start with the read factory... if there are enough samples, read them in */
2008-07-11 20:03:55 +00:00
if ( usable_read ) {
2007-08-21 15:51:49 +00:00
if ( ast_slinfactory_read ( & audiohook - > read_factory , buf1 , samples ) ) {
2007-08-08 19:30:52 +00:00
read_buf = buf1 ;
2007-08-21 15:51:49 +00:00
/* Adjust read volume if need be */
if ( audiohook - > options . read_volume ) {
int count = 0 ;
short adjust_value = abs ( audiohook - > options . read_volume ) ;
for ( count = 0 ; count < samples ; count + + ) {
if ( audiohook - > options . read_volume > 0 )
ast_slinear_saturated_multiply ( & buf1 [ count ] , & adjust_value ) ;
else if ( audiohook - > options . read_volume < 0 )
ast_slinear_saturated_divide ( & buf1 [ count ] , & adjust_value ) ;
}
2007-08-08 19:30:52 +00:00
}
}
2011-02-04 16:55:39 +00:00
}
ast_debug ( 1 , " Failed to get %d samples from read factory %p \n " , ( int ) samples , & audiohook - > read_factory ) ;
2007-08-08 19:30:52 +00:00
/* Move on to the write factory... if there are enough samples, read them in */
2008-07-11 20:03:55 +00:00
if ( usable_write ) {
2007-08-21 15:51:49 +00:00
if ( ast_slinfactory_read ( & audiohook - > write_factory , buf2 , samples ) ) {
2007-08-08 19:30:52 +00:00
write_buf = buf2 ;
2007-08-21 15:51:49 +00:00
/* Adjust write volume if need be */
if ( audiohook - > options . write_volume ) {
int count = 0 ;
short adjust_value = abs ( audiohook - > options . write_volume ) ;
for ( count = 0 ; count < samples ; count + + ) {
if ( audiohook - > options . write_volume > 0 )
ast_slinear_saturated_multiply ( & buf2 [ count ] , & adjust_value ) ;
else if ( audiohook - > options . write_volume < 0 )
ast_slinear_saturated_divide ( & buf2 [ count ] , & adjust_value ) ;
}
2007-08-08 19:30:52 +00:00
}
}
2011-02-04 16:55:39 +00:00
}
ast_debug ( 1 , " Failed to get %d samples from write factory %p \n " , ( int ) samples , & audiohook - > write_factory ) ;
2007-08-08 19:30:52 +00:00
/* Basically we figure out which buffer to use... and if mixing can be done here */
2011-03-14 13:12:51 +00:00
if ( read_buf & & read_reference ) {
frame . data . ptr = buf1 ;
2011-03-11 18:54:45 +00:00
* read_reference = ast_frdup ( & frame ) ;
}
2011-03-14 13:12:51 +00:00
if ( write_buf & & write_reference ) {
frame . data . ptr = buf2 ;
2011-03-11 18:54:45 +00:00
* write_reference = ast_frdup ( & frame ) ;
}
if ( read_buf & & write_buf ) {
2011-03-14 13:12:51 +00:00
for ( i = 0 , data1 = read_buf , data2 = write_buf ; i < samples ; i + + , data1 + + , data2 + + ) {
2007-08-08 19:30:52 +00:00
ast_slinear_saturated_add ( data1 , data2 ) ;
2011-03-14 13:12:51 +00:00
}
final_buf = buf1 ;
} else if ( read_buf ) {
2007-08-08 19:30:52 +00:00
final_buf = buf1 ;
2011-03-14 13:12:51 +00:00
} else if ( write_buf ) {
final_buf = buf2 ;
} else {
return NULL ;
2011-03-11 18:54:45 +00:00
}
2007-08-08 19:30:52 +00:00
/* Make the final buffer part of the frame, so it gets duplicated fine */
2008-05-22 16:29:54 +00:00
frame . data . ptr = final_buf ;
2007-08-08 19:30:52 +00:00
/* Yahoo, a combined copy of the audio! */
return ast_frdup ( & frame ) ;
}
2011-03-11 18:54:45 +00:00
static struct ast_frame * audiohook_read_frame_helper ( struct ast_audiohook * audiohook , size_t samples , enum ast_audiohook_direction direction , struct ast_format * format , struct ast_frame * * read_reference , struct ast_frame * * write_reference )
2007-08-08 19:30:52 +00:00
{
struct ast_frame * read_frame = NULL , * final_frame = NULL ;
2011-02-03 16:22:10 +00:00
struct ast_format tmp_fmt ;
Media Project Phase2: SILK 8khz-24khz, SLINEAR 8khz-192khz, SPEEX 32khz, hd audio ConfBridge, and other stuff
-Functional changes
1. Dynamic global format list build by codecs defined in codecs.conf
2. SILK 8khz, 12khz, 16khz, and 24khz with custom attributes defined in codecs.conf
3. Negotiation of SILK attributes in chan_sip.
4. SPEEX 32khz with translation
5. SLINEAR 8khz, 12khz, 24khz, 32khz, 44.1khz, 48khz, 96khz, 192khz with translation
using codec_resample.c
6. Various changes to RTP code required to properly handle the dynamic format list
and formats with attributes.
7. ConfBridge now dynamically jumps to the best possible sample rate. This allows
for conferences to take advantage of HD audio (Which sounds awesome)
8. Audiohooks are no longer limited to 8khz audio, and most effects have been
updated to take advantage of this such as Volume, DENOISE, PITCH_SHIFT.
9. codec_resample now uses its own code rather than depending on libresample.
-Organizational changes
Global format list is moved from frame.c to format.c
Various format specific functions moved from frame.c to format.c
Review: https://reviewboard.asterisk.org/r/1104/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308582 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-22 23:04:49 +00:00
int samples_converted ;
/* the number of samples requested is based on the format they are requesting. Inorder
* to process this correctly samples must be converted to our internal sample rate */
if ( audiohook - > hook_internal_samp_rate = = ast_format_rate ( format ) ) {
samples_converted = samples ;
} else if ( audiohook - > hook_internal_samp_rate > ast_format_rate ( format ) ) {
samples_converted = samples * ( audiohook - > hook_internal_samp_rate / ( float ) ast_format_rate ( format ) ) ;
} else {
samples_converted = samples * ( ast_format_rate ( format ) / ( float ) audiohook - > hook_internal_samp_rate ) ;
}
2007-08-08 19:30:52 +00:00
2011-03-11 18:54:45 +00:00
if ( ! ( read_frame = ( direction = = AST_AUDIOHOOK_DIRECTION_BOTH ?
audiohook_read_frame_both ( audiohook , samples_converted , read_reference , write_reference ) :
audiohook_read_frame_single ( audiohook , samples_converted , direction ) ) ) ) {
return NULL ;
Media Project Phase2: SILK 8khz-24khz, SLINEAR 8khz-192khz, SPEEX 32khz, hd audio ConfBridge, and other stuff
-Functional changes
1. Dynamic global format list build by codecs defined in codecs.conf
2. SILK 8khz, 12khz, 16khz, and 24khz with custom attributes defined in codecs.conf
3. Negotiation of SILK attributes in chan_sip.
4. SPEEX 32khz with translation
5. SLINEAR 8khz, 12khz, 24khz, 32khz, 44.1khz, 48khz, 96khz, 192khz with translation
using codec_resample.c
6. Various changes to RTP code required to properly handle the dynamic format list
and formats with attributes.
7. ConfBridge now dynamically jumps to the best possible sample rate. This allows
for conferences to take advantage of HD audio (Which sounds awesome)
8. Audiohooks are no longer limited to 8khz audio, and most effects have been
updated to take advantage of this such as Volume, DENOISE, PITCH_SHIFT.
9. codec_resample now uses its own code rather than depending on libresample.
-Organizational changes
Global format list is moved from frame.c to format.c
Various format specific functions moved from frame.c to format.c
Review: https://reviewboard.asterisk.org/r/1104/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308582 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-22 23:04:49 +00:00
}
2007-08-08 19:30:52 +00:00
/* If they don't want signed linear back out, we'll have to send it through the translation path */
Media Project Phase2: SILK 8khz-24khz, SLINEAR 8khz-192khz, SPEEX 32khz, hd audio ConfBridge, and other stuff
-Functional changes
1. Dynamic global format list build by codecs defined in codecs.conf
2. SILK 8khz, 12khz, 16khz, and 24khz with custom attributes defined in codecs.conf
3. Negotiation of SILK attributes in chan_sip.
4. SPEEX 32khz with translation
5. SLINEAR 8khz, 12khz, 24khz, 32khz, 44.1khz, 48khz, 96khz, 192khz with translation
using codec_resample.c
6. Various changes to RTP code required to properly handle the dynamic format list
and formats with attributes.
7. ConfBridge now dynamically jumps to the best possible sample rate. This allows
for conferences to take advantage of HD audio (Which sounds awesome)
8. Audiohooks are no longer limited to 8khz audio, and most effects have been
updated to take advantage of this such as Volume, DENOISE, PITCH_SHIFT.
9. codec_resample now uses its own code rather than depending on libresample.
-Organizational changes
Global format list is moved from frame.c to format.c
Various format specific functions moved from frame.c to format.c
Review: https://reviewboard.asterisk.org/r/1104/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308582 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-22 23:04:49 +00:00
if ( format - > id ! = ast_format_slin_by_rate ( audiohook - > hook_internal_samp_rate ) ) {
2007-08-08 19:30:52 +00:00
/* Rebuild translation path if different format then previously */
2011-02-03 16:22:10 +00:00
if ( ast_format_cmp ( format , & audiohook - > format ) = = AST_FORMAT_CMP_NOT_EQUAL ) {
2007-08-08 19:30:52 +00:00
if ( audiohook - > trans_pvt ) {
ast_translator_free_path ( audiohook - > trans_pvt ) ;
audiohook - > trans_pvt = NULL ;
}
Media Project Phase2: SILK 8khz-24khz, SLINEAR 8khz-192khz, SPEEX 32khz, hd audio ConfBridge, and other stuff
-Functional changes
1. Dynamic global format list build by codecs defined in codecs.conf
2. SILK 8khz, 12khz, 16khz, and 24khz with custom attributes defined in codecs.conf
3. Negotiation of SILK attributes in chan_sip.
4. SPEEX 32khz with translation
5. SLINEAR 8khz, 12khz, 24khz, 32khz, 44.1khz, 48khz, 96khz, 192khz with translation
using codec_resample.c
6. Various changes to RTP code required to properly handle the dynamic format list
and formats with attributes.
7. ConfBridge now dynamically jumps to the best possible sample rate. This allows
for conferences to take advantage of HD audio (Which sounds awesome)
8. Audiohooks are no longer limited to 8khz audio, and most effects have been
updated to take advantage of this such as Volume, DENOISE, PITCH_SHIFT.
9. codec_resample now uses its own code rather than depending on libresample.
-Organizational changes
Global format list is moved from frame.c to format.c
Various format specific functions moved from frame.c to format.c
Review: https://reviewboard.asterisk.org/r/1104/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308582 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-22 23:04:49 +00:00
2007-08-08 19:30:52 +00:00
/* Setup new translation path for this format... if we fail we can't very well return signed linear so free the frame and return nothing */
Media Project Phase2: SILK 8khz-24khz, SLINEAR 8khz-192khz, SPEEX 32khz, hd audio ConfBridge, and other stuff
-Functional changes
1. Dynamic global format list build by codecs defined in codecs.conf
2. SILK 8khz, 12khz, 16khz, and 24khz with custom attributes defined in codecs.conf
3. Negotiation of SILK attributes in chan_sip.
4. SPEEX 32khz with translation
5. SLINEAR 8khz, 12khz, 24khz, 32khz, 44.1khz, 48khz, 96khz, 192khz with translation
using codec_resample.c
6. Various changes to RTP code required to properly handle the dynamic format list
and formats with attributes.
7. ConfBridge now dynamically jumps to the best possible sample rate. This allows
for conferences to take advantage of HD audio (Which sounds awesome)
8. Audiohooks are no longer limited to 8khz audio, and most effects have been
updated to take advantage of this such as Volume, DENOISE, PITCH_SHIFT.
9. codec_resample now uses its own code rather than depending on libresample.
-Organizational changes
Global format list is moved from frame.c to format.c
Various format specific functions moved from frame.c to format.c
Review: https://reviewboard.asterisk.org/r/1104/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308582 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-22 23:04:49 +00:00
if ( ! ( audiohook - > trans_pvt = ast_translator_build_path ( format , ast_format_set ( & tmp_fmt , ast_format_slin_by_rate ( audiohook - > hook_internal_samp_rate ) , 0 ) ) ) ) {
2007-08-08 19:30:52 +00:00
ast_frfree ( read_frame ) ;
return NULL ;
}
Media Project Phase2: SILK 8khz-24khz, SLINEAR 8khz-192khz, SPEEX 32khz, hd audio ConfBridge, and other stuff
-Functional changes
1. Dynamic global format list build by codecs defined in codecs.conf
2. SILK 8khz, 12khz, 16khz, and 24khz with custom attributes defined in codecs.conf
3. Negotiation of SILK attributes in chan_sip.
4. SPEEX 32khz with translation
5. SLINEAR 8khz, 12khz, 24khz, 32khz, 44.1khz, 48khz, 96khz, 192khz with translation
using codec_resample.c
6. Various changes to RTP code required to properly handle the dynamic format list
and formats with attributes.
7. ConfBridge now dynamically jumps to the best possible sample rate. This allows
for conferences to take advantage of HD audio (Which sounds awesome)
8. Audiohooks are no longer limited to 8khz audio, and most effects have been
updated to take advantage of this such as Volume, DENOISE, PITCH_SHIFT.
9. codec_resample now uses its own code rather than depending on libresample.
-Organizational changes
Global format list is moved from frame.c to format.c
Various format specific functions moved from frame.c to format.c
Review: https://reviewboard.asterisk.org/r/1104/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308582 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-22 23:04:49 +00:00
ast_format_copy ( & audiohook - > format , format ) ;
2007-08-08 19:30:52 +00:00
}
/* Convert to requested format, and allow the read in frame to be freed */
final_frame = ast_translate ( audiohook - > trans_pvt , read_frame , 1 ) ;
} else {
final_frame = read_frame ;
}
return final_frame ;
}
2011-03-11 18:54:45 +00:00
/*! \brief Reads a frame in from the audiohook structure
* \ param audiohook Audiohook structure
* \ param samples Number of samples wanted in requested output format
* \ param direction Direction the audio frame came from
* \ param format Format of frame remote side wants back
* \ return Returns frame on success , NULL on failure
*/
struct ast_frame * ast_audiohook_read_frame ( struct ast_audiohook * audiohook , size_t samples , enum ast_audiohook_direction direction , struct ast_format * format )
{
return audiohook_read_frame_helper ( audiohook , samples , direction , format , NULL , NULL ) ;
}
/*! \brief Reads a frame in from the audiohook structure
* \ param audiohook Audiohook structure
* \ param samples Number of samples wanted
* \ param direction Direction the audio frame came from
* \ param format Format of frame remote side wants back
* \ param read_frame frame pointer for copying read frame data
* \ param write_frame frame pointer for copying write frame data
* \ return Returns frame on success , NULL on failure
*/
struct ast_frame * ast_audiohook_read_frame_all ( struct ast_audiohook * audiohook , size_t samples , struct ast_format * format , struct ast_frame * * read_frame , struct ast_frame * * write_frame )
{
return audiohook_read_frame_helper ( audiohook , samples , AST_AUDIOHOOK_DIRECTION_BOTH , format , read_frame , write_frame ) ;
}
Media Project Phase2: SILK 8khz-24khz, SLINEAR 8khz-192khz, SPEEX 32khz, hd audio ConfBridge, and other stuff
-Functional changes
1. Dynamic global format list build by codecs defined in codecs.conf
2. SILK 8khz, 12khz, 16khz, and 24khz with custom attributes defined in codecs.conf
3. Negotiation of SILK attributes in chan_sip.
4. SPEEX 32khz with translation
5. SLINEAR 8khz, 12khz, 24khz, 32khz, 44.1khz, 48khz, 96khz, 192khz with translation
using codec_resample.c
6. Various changes to RTP code required to properly handle the dynamic format list
and formats with attributes.
7. ConfBridge now dynamically jumps to the best possible sample rate. This allows
for conferences to take advantage of HD audio (Which sounds awesome)
8. Audiohooks are no longer limited to 8khz audio, and most effects have been
updated to take advantage of this such as Volume, DENOISE, PITCH_SHIFT.
9. codec_resample now uses its own code rather than depending on libresample.
-Organizational changes
Global format list is moved from frame.c to format.c
Various format specific functions moved from frame.c to format.c
Review: https://reviewboard.asterisk.org/r/1104/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308582 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-22 23:04:49 +00:00
static void audiohook_list_set_samplerate_compatibility ( struct ast_audiohook_list * audiohook_list )
{
struct ast_audiohook * ah = NULL ;
audiohook_list - > native_slin_compatible = 1 ;
AST_LIST_TRAVERSE ( & audiohook_list - > manipulate_list , ah , list ) {
if ( ! ( ah - > init_flags & AST_AUDIOHOOK_MANIPULATE_ALL_RATES ) ) {
audiohook_list - > native_slin_compatible = 0 ;
return ;
}
}
}
2007-08-08 19:30:52 +00:00
/*! \brief Attach audiohook to channel
* \ param chan Channel
* \ param audiohook Audiohook structure
* \ return Returns 0 on success , - 1 on failure
*/
int ast_audiohook_attach ( struct ast_channel * chan , struct ast_audiohook * audiohook )
{
ast_channel_lock ( chan ) ;
2012-02-20 23:43:27 +00:00
if ( ! ast_channel_audiohooks ( chan ) ) {
struct ast_audiohook_list * ahlist ;
2007-08-08 19:30:52 +00:00
/* Whoops... allocate a new structure */
2012-02-20 23:43:27 +00:00
if ( ! ( ahlist = ast_calloc ( 1 , sizeof ( * ahlist ) ) ) ) {
2007-08-08 19:30:52 +00:00
ast_channel_unlock ( chan ) ;
return - 1 ;
}
2012-02-20 23:43:27 +00:00
ast_channel_audiohooks_set ( chan , ahlist ) ;
AST_LIST_HEAD_INIT_NOLOCK ( & ast_channel_audiohooks ( chan ) - > spy_list ) ;
AST_LIST_HEAD_INIT_NOLOCK ( & ast_channel_audiohooks ( chan ) - > whisper_list ) ;
AST_LIST_HEAD_INIT_NOLOCK ( & ast_channel_audiohooks ( chan ) - > manipulate_list ) ;
Media Project Phase2: SILK 8khz-24khz, SLINEAR 8khz-192khz, SPEEX 32khz, hd audio ConfBridge, and other stuff
-Functional changes
1. Dynamic global format list build by codecs defined in codecs.conf
2. SILK 8khz, 12khz, 16khz, and 24khz with custom attributes defined in codecs.conf
3. Negotiation of SILK attributes in chan_sip.
4. SPEEX 32khz with translation
5. SLINEAR 8khz, 12khz, 24khz, 32khz, 44.1khz, 48khz, 96khz, 192khz with translation
using codec_resample.c
6. Various changes to RTP code required to properly handle the dynamic format list
and formats with attributes.
7. ConfBridge now dynamically jumps to the best possible sample rate. This allows
for conferences to take advantage of HD audio (Which sounds awesome)
8. Audiohooks are no longer limited to 8khz audio, and most effects have been
updated to take advantage of this such as Volume, DENOISE, PITCH_SHIFT.
9. codec_resample now uses its own code rather than depending on libresample.
-Organizational changes
Global format list is moved from frame.c to format.c
Various format specific functions moved from frame.c to format.c
Review: https://reviewboard.asterisk.org/r/1104/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308582 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-22 23:04:49 +00:00
/* This sample rate will adjust as necessary when writing to the list. */
2012-02-20 23:43:27 +00:00
ast_channel_audiohooks ( chan ) - > list_internal_samp_rate = 8000 ;
2007-08-08 19:30:52 +00:00
}
/* Drop into respective list */
if ( audiohook - > type = = AST_AUDIOHOOK_TYPE_SPY )
2012-02-20 23:43:27 +00:00
AST_LIST_INSERT_TAIL ( & ast_channel_audiohooks ( chan ) - > spy_list , audiohook , list ) ;
2007-08-08 19:30:52 +00:00
else if ( audiohook - > type = = AST_AUDIOHOOK_TYPE_WHISPER )
2012-02-20 23:43:27 +00:00
AST_LIST_INSERT_TAIL ( & ast_channel_audiohooks ( chan ) - > whisper_list , audiohook , list ) ;
2007-08-08 19:30:52 +00:00
else if ( audiohook - > type = = AST_AUDIOHOOK_TYPE_MANIPULATE )
2012-02-20 23:43:27 +00:00
AST_LIST_INSERT_TAIL ( & ast_channel_audiohooks ( chan ) - > manipulate_list , audiohook , list ) ;
2007-08-08 19:30:52 +00:00
Media Project Phase2: SILK 8khz-24khz, SLINEAR 8khz-192khz, SPEEX 32khz, hd audio ConfBridge, and other stuff
-Functional changes
1. Dynamic global format list build by codecs defined in codecs.conf
2. SILK 8khz, 12khz, 16khz, and 24khz with custom attributes defined in codecs.conf
3. Negotiation of SILK attributes in chan_sip.
4. SPEEX 32khz with translation
5. SLINEAR 8khz, 12khz, 24khz, 32khz, 44.1khz, 48khz, 96khz, 192khz with translation
using codec_resample.c
6. Various changes to RTP code required to properly handle the dynamic format list
and formats with attributes.
7. ConfBridge now dynamically jumps to the best possible sample rate. This allows
for conferences to take advantage of HD audio (Which sounds awesome)
8. Audiohooks are no longer limited to 8khz audio, and most effects have been
updated to take advantage of this such as Volume, DENOISE, PITCH_SHIFT.
9. codec_resample now uses its own code rather than depending on libresample.
-Organizational changes
Global format list is moved from frame.c to format.c
Various format specific functions moved from frame.c to format.c
Review: https://reviewboard.asterisk.org/r/1104/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308582 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-22 23:04:49 +00:00
2012-02-20 23:43:27 +00:00
audiohook_set_internal_rate ( audiohook , ast_channel_audiohooks ( chan ) - > list_internal_samp_rate , 1 ) ;
audiohook_list_set_samplerate_compatibility ( ast_channel_audiohooks ( chan ) ) ;
Media Project Phase2: SILK 8khz-24khz, SLINEAR 8khz-192khz, SPEEX 32khz, hd audio ConfBridge, and other stuff
-Functional changes
1. Dynamic global format list build by codecs defined in codecs.conf
2. SILK 8khz, 12khz, 16khz, and 24khz with custom attributes defined in codecs.conf
3. Negotiation of SILK attributes in chan_sip.
4. SPEEX 32khz with translation
5. SLINEAR 8khz, 12khz, 24khz, 32khz, 44.1khz, 48khz, 96khz, 192khz with translation
using codec_resample.c
6. Various changes to RTP code required to properly handle the dynamic format list
and formats with attributes.
7. ConfBridge now dynamically jumps to the best possible sample rate. This allows
for conferences to take advantage of HD audio (Which sounds awesome)
8. Audiohooks are no longer limited to 8khz audio, and most effects have been
updated to take advantage of this such as Volume, DENOISE, PITCH_SHIFT.
9. codec_resample now uses its own code rather than depending on libresample.
-Organizational changes
Global format list is moved from frame.c to format.c
Various format specific functions moved from frame.c to format.c
Review: https://reviewboard.asterisk.org/r/1104/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308582 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-22 23:04:49 +00:00
2007-08-08 19:30:52 +00:00
/* Change status over to running since it is now attached */
2009-11-20 17:26:20 +00:00
ast_audiohook_update_status ( audiohook , AST_AUDIOHOOK_STATUS_RUNNING ) ;
2007-08-08 19:30:52 +00:00
ast_channel_unlock ( chan ) ;
return 0 ;
}
2009-11-20 17:26:20 +00:00
/*! \brief Update audiohook's status
* \ param audiohook Audiohook structure
2010-06-08 14:38:18 +00:00
* \ param status Audiohook status enum
2010-01-08 19:39:30 +00:00
*
* \ note once status is updated to DONE , this function can not be used to set the
* status back to any other setting . Setting DONE effectively locks the status as such .
2009-11-20 17:26:20 +00:00
*/
2010-01-08 19:39:30 +00:00
2009-11-20 17:26:20 +00:00
void ast_audiohook_update_status ( struct ast_audiohook * audiohook , enum ast_audiohook_status status )
{
ast_audiohook_lock ( audiohook ) ;
if ( audiohook - > status ! = AST_AUDIOHOOK_STATUS_DONE ) {
audiohook - > status = status ;
ast_cond_signal ( & audiohook - > trigger ) ;
}
ast_audiohook_unlock ( audiohook ) ;
}
2007-08-08 19:30:52 +00:00
/*! \brief Detach audiohook from channel
* \ param audiohook Audiohook structure
* \ return Returns 0 on success , - 1 on failure
*/
int ast_audiohook_detach ( struct ast_audiohook * audiohook )
{
2009-02-06 15:59:17 +00:00
if ( audiohook - > status = = AST_AUDIOHOOK_STATUS_NEW | | audiohook - > status = = AST_AUDIOHOOK_STATUS_DONE )
2007-08-08 19:30:52 +00:00
return 0 ;
2009-11-20 17:26:20 +00:00
ast_audiohook_update_status ( audiohook , AST_AUDIOHOOK_STATUS_SHUTDOWN ) ;
2007-08-08 19:30:52 +00:00
while ( audiohook - > status ! = AST_AUDIOHOOK_STATUS_DONE )
ast_audiohook_trigger_wait ( audiohook ) ;
return 0 ;
}
/*! \brief Detach audiohooks from list and destroy said list
* \ param audiohook_list List of audiohooks
* \ return Returns 0 on success , - 1 on failure
*/
int ast_audiohook_detach_list ( struct ast_audiohook_list * audiohook_list )
{
int i = 0 ;
struct ast_audiohook * audiohook = NULL ;
/* Drop any spies */
2007-11-08 05:28:47 +00:00
while ( ( audiohook = AST_LIST_REMOVE_HEAD ( & audiohook_list - > spy_list , list ) ) ) {
2009-11-20 17:26:20 +00:00
ast_audiohook_update_status ( audiohook , AST_AUDIOHOOK_STATUS_DONE ) ;
2007-08-08 19:30:52 +00:00
}
/* Drop any whispering sources */
2007-11-08 05:28:47 +00:00
while ( ( audiohook = AST_LIST_REMOVE_HEAD ( & audiohook_list - > whisper_list , list ) ) ) {
2009-11-20 17:26:20 +00:00
ast_audiohook_update_status ( audiohook , AST_AUDIOHOOK_STATUS_DONE ) ;
2007-08-08 19:30:52 +00:00
}
/* Drop any manipulaters */
2007-11-08 05:28:47 +00:00
while ( ( audiohook = AST_LIST_REMOVE_HEAD ( & audiohook_list - > manipulate_list , list ) ) ) {
2009-11-20 17:26:20 +00:00
ast_audiohook_update_status ( audiohook , AST_AUDIOHOOK_STATUS_DONE ) ;
2007-08-08 19:30:52 +00:00
audiohook - > manipulate_callback ( audiohook , NULL , NULL , 0 ) ;
}
/* Drop translation paths if present */
for ( i = 0 ; i < 2 ; i + + ) {
if ( audiohook_list - > in_translate [ i ] . trans_pvt )
ast_translator_free_path ( audiohook_list - > in_translate [ i ] . trans_pvt ) ;
if ( audiohook_list - > out_translate [ i ] . trans_pvt )
ast_translator_free_path ( audiohook_list - > out_translate [ i ] . trans_pvt ) ;
}
/* Free ourselves */
ast_free ( audiohook_list ) ;
return 0 ;
}
2008-12-19 22:26:16 +00:00
/*! \brief find an audiohook based on its source
* \ param audiohook_list The list of audiohooks to search in
* \ param source The source of the audiohook we wish to find
* \ return Return the corresponding audiohook or NULL if it cannot be found .
*/
2007-08-08 19:30:52 +00:00
static struct ast_audiohook * find_audiohook_by_source ( struct ast_audiohook_list * audiohook_list , const char * source )
{
struct ast_audiohook * audiohook = NULL ;
AST_LIST_TRAVERSE ( & audiohook_list - > spy_list , audiohook , list ) {
if ( ! strcasecmp ( audiohook - > source , source ) )
return audiohook ;
}
AST_LIST_TRAVERSE ( & audiohook_list - > whisper_list , audiohook , list ) {
if ( ! strcasecmp ( audiohook - > source , source ) )
return audiohook ;
}
AST_LIST_TRAVERSE ( & audiohook_list - > manipulate_list , audiohook , list ) {
if ( ! strcasecmp ( audiohook - > source , source ) )
return audiohook ;
}
return NULL ;
}
2008-12-19 23:45:00 +00:00
void ast_audiohook_move_by_source ( struct ast_channel * old_chan , struct ast_channel * new_chan , const char * source )
2008-12-19 22:26:16 +00:00
{
2009-03-31 14:07:36 +00:00
struct ast_audiohook * audiohook ;
2010-01-08 19:39:30 +00:00
enum ast_audiohook_status oldstatus ;
2008-12-19 22:26:16 +00:00
2012-02-20 23:43:27 +00:00
if ( ! ast_channel_audiohooks ( old_chan ) | | ! ( audiohook = find_audiohook_by_source ( ast_channel_audiohooks ( old_chan ) , source ) ) ) {
2008-12-19 22:26:16 +00:00
return ;
}
2009-03-31 14:07:36 +00:00
2008-12-19 22:26:16 +00:00
/* By locking both channels and the audiohook, we can assure that
* another thread will not have a chance to read the audiohook ' s status
* as done , even though ast_audiohook_remove signals the trigger
2010-01-08 19:39:30 +00:00
* condition .
2008-12-19 22:26:16 +00:00
*/
ast_audiohook_lock ( audiohook ) ;
2010-01-08 19:39:30 +00:00
oldstatus = audiohook - > status ;
2008-12-19 22:26:16 +00:00
ast_audiohook_remove ( old_chan , audiohook ) ;
ast_audiohook_attach ( new_chan , audiohook ) ;
2010-01-08 19:39:30 +00:00
audiohook - > status = oldstatus ;
2008-12-19 22:26:16 +00:00
ast_audiohook_unlock ( audiohook ) ;
}
2007-08-08 19:30:52 +00:00
/*! \brief Detach specified source audiohook from channel
* \ param chan Channel to detach from
* \ param source Name of source to detach
* \ return Returns 0 on success , - 1 on failure
*/
int ast_audiohook_detach_source ( struct ast_channel * chan , const char * source )
{
struct ast_audiohook * audiohook = NULL ;
ast_channel_lock ( chan ) ;
/* Ensure the channel has audiohooks on it */
2012-02-20 23:43:27 +00:00
if ( ! ast_channel_audiohooks ( chan ) ) {
2007-08-08 19:30:52 +00:00
ast_channel_unlock ( chan ) ;
return - 1 ;
}
2012-02-20 23:43:27 +00:00
audiohook = find_audiohook_by_source ( ast_channel_audiohooks ( chan ) , source ) ;
2007-08-08 19:30:52 +00:00
ast_channel_unlock ( chan ) ;
if ( audiohook & & audiohook - > status ! = AST_AUDIOHOOK_STATUS_DONE )
2009-11-20 17:26:20 +00:00
ast_audiohook_update_status ( audiohook , AST_AUDIOHOOK_STATUS_SHUTDOWN ) ;
2007-08-08 19:30:52 +00:00
return ( audiohook ? 0 : - 1 ) ;
}
2008-05-01 16:57:19 +00:00
/*!
* \ brief Remove an audiohook from a specified channel
*
* \ param chan Channel to remove from
* \ param audiohook Audiohook to remove
*
* \ return Returns 0 on success , - 1 on failure
*
* \ note The channel does not need to be locked before calling this function
*/
int ast_audiohook_remove ( struct ast_channel * chan , struct ast_audiohook * audiohook )
{
ast_channel_lock ( chan ) ;
2012-02-20 23:43:27 +00:00
if ( ! ast_channel_audiohooks ( chan ) ) {
2008-05-01 16:57:19 +00:00
ast_channel_unlock ( chan ) ;
return - 1 ;
}
if ( audiohook - > type = = AST_AUDIOHOOK_TYPE_SPY )
2012-02-20 23:43:27 +00:00
AST_LIST_REMOVE ( & ast_channel_audiohooks ( chan ) - > spy_list , audiohook , list ) ;
2008-05-01 16:57:19 +00:00
else if ( audiohook - > type = = AST_AUDIOHOOK_TYPE_WHISPER )
2012-02-20 23:43:27 +00:00
AST_LIST_REMOVE ( & ast_channel_audiohooks ( chan ) - > whisper_list , audiohook , list ) ;
2008-05-01 16:57:19 +00:00
else if ( audiohook - > type = = AST_AUDIOHOOK_TYPE_MANIPULATE )
2012-02-20 23:43:27 +00:00
AST_LIST_REMOVE ( & ast_channel_audiohooks ( chan ) - > manipulate_list , audiohook , list ) ;
2008-05-01 16:57:19 +00:00
2012-02-20 23:43:27 +00:00
audiohook_list_set_samplerate_compatibility ( ast_channel_audiohooks ( chan ) ) ;
2009-11-20 17:26:20 +00:00
ast_audiohook_update_status ( audiohook , AST_AUDIOHOOK_STATUS_DONE ) ;
2008-05-01 16:57:19 +00:00
ast_channel_unlock ( chan ) ;
return 0 ;
}
2007-08-08 19:30:52 +00:00
/*! \brief Pass a DTMF frame off to be handled by the audiohook core
* \ param chan Channel that the list is coming off of
* \ param audiohook_list List of audiohooks
* \ param direction Direction frame is coming in from
* \ param frame The frame itself
* \ return Return frame on success , NULL on failure
*/
static struct ast_frame * dtmf_audiohook_write_list ( struct ast_channel * chan , struct ast_audiohook_list * audiohook_list , enum ast_audiohook_direction direction , struct ast_frame * frame )
{
struct ast_audiohook * audiohook = NULL ;
Media Project Phase2: SILK 8khz-24khz, SLINEAR 8khz-192khz, SPEEX 32khz, hd audio ConfBridge, and other stuff
-Functional changes
1. Dynamic global format list build by codecs defined in codecs.conf
2. SILK 8khz, 12khz, 16khz, and 24khz with custom attributes defined in codecs.conf
3. Negotiation of SILK attributes in chan_sip.
4. SPEEX 32khz with translation
5. SLINEAR 8khz, 12khz, 24khz, 32khz, 44.1khz, 48khz, 96khz, 192khz with translation
using codec_resample.c
6. Various changes to RTP code required to properly handle the dynamic format list
and formats with attributes.
7. ConfBridge now dynamically jumps to the best possible sample rate. This allows
for conferences to take advantage of HD audio (Which sounds awesome)
8. Audiohooks are no longer limited to 8khz audio, and most effects have been
updated to take advantage of this such as Volume, DENOISE, PITCH_SHIFT.
9. codec_resample now uses its own code rather than depending on libresample.
-Organizational changes
Global format list is moved from frame.c to format.c
Various format specific functions moved from frame.c to format.c
Review: https://reviewboard.asterisk.org/r/1104/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308582 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-22 23:04:49 +00:00
int removed = 0 ;
2007-08-08 19:30:52 +00:00
AST_LIST_TRAVERSE_SAFE_BEGIN ( & audiohook_list - > manipulate_list , audiohook , list ) {
ast_audiohook_lock ( audiohook ) ;
if ( audiohook - > status ! = AST_AUDIOHOOK_STATUS_RUNNING ) {
2007-11-08 05:28:47 +00:00
AST_LIST_REMOVE_CURRENT ( list ) ;
Media Project Phase2: SILK 8khz-24khz, SLINEAR 8khz-192khz, SPEEX 32khz, hd audio ConfBridge, and other stuff
-Functional changes
1. Dynamic global format list build by codecs defined in codecs.conf
2. SILK 8khz, 12khz, 16khz, and 24khz with custom attributes defined in codecs.conf
3. Negotiation of SILK attributes in chan_sip.
4. SPEEX 32khz with translation
5. SLINEAR 8khz, 12khz, 24khz, 32khz, 44.1khz, 48khz, 96khz, 192khz with translation
using codec_resample.c
6. Various changes to RTP code required to properly handle the dynamic format list
and formats with attributes.
7. ConfBridge now dynamically jumps to the best possible sample rate. This allows
for conferences to take advantage of HD audio (Which sounds awesome)
8. Audiohooks are no longer limited to 8khz audio, and most effects have been
updated to take advantage of this such as Volume, DENOISE, PITCH_SHIFT.
9. codec_resample now uses its own code rather than depending on libresample.
-Organizational changes
Global format list is moved from frame.c to format.c
Various format specific functions moved from frame.c to format.c
Review: https://reviewboard.asterisk.org/r/1104/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308582 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-22 23:04:49 +00:00
removed = 1 ;
2009-11-20 17:26:20 +00:00
ast_audiohook_update_status ( audiohook , AST_AUDIOHOOK_STATUS_DONE ) ;
2007-08-08 19:30:52 +00:00
ast_audiohook_unlock ( audiohook ) ;
audiohook - > manipulate_callback ( audiohook , NULL , NULL , 0 ) ;
continue ;
}
if ( ast_test_flag ( audiohook , AST_AUDIOHOOK_WANTS_DTMF ) )
audiohook - > manipulate_callback ( audiohook , chan , frame , direction ) ;
ast_audiohook_unlock ( audiohook ) ;
}
2007-11-08 05:28:47 +00:00
AST_LIST_TRAVERSE_SAFE_END ;
2007-08-08 19:30:52 +00:00
Media Project Phase2: SILK 8khz-24khz, SLINEAR 8khz-192khz, SPEEX 32khz, hd audio ConfBridge, and other stuff
-Functional changes
1. Dynamic global format list build by codecs defined in codecs.conf
2. SILK 8khz, 12khz, 16khz, and 24khz with custom attributes defined in codecs.conf
3. Negotiation of SILK attributes in chan_sip.
4. SPEEX 32khz with translation
5. SLINEAR 8khz, 12khz, 24khz, 32khz, 44.1khz, 48khz, 96khz, 192khz with translation
using codec_resample.c
6. Various changes to RTP code required to properly handle the dynamic format list
and formats with attributes.
7. ConfBridge now dynamically jumps to the best possible sample rate. This allows
for conferences to take advantage of HD audio (Which sounds awesome)
8. Audiohooks are no longer limited to 8khz audio, and most effects have been
updated to take advantage of this such as Volume, DENOISE, PITCH_SHIFT.
9. codec_resample now uses its own code rather than depending on libresample.
-Organizational changes
Global format list is moved from frame.c to format.c
Various format specific functions moved from frame.c to format.c
Review: https://reviewboard.asterisk.org/r/1104/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308582 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-22 23:04:49 +00:00
/* if an audiohook got removed, reset samplerate compatibility */
if ( removed ) {
audiohook_list_set_samplerate_compatibility ( audiohook_list ) ;
}
2007-08-08 19:30:52 +00:00
return frame ;
}
Media Project Phase2: SILK 8khz-24khz, SLINEAR 8khz-192khz, SPEEX 32khz, hd audio ConfBridge, and other stuff
-Functional changes
1. Dynamic global format list build by codecs defined in codecs.conf
2. SILK 8khz, 12khz, 16khz, and 24khz with custom attributes defined in codecs.conf
3. Negotiation of SILK attributes in chan_sip.
4. SPEEX 32khz with translation
5. SLINEAR 8khz, 12khz, 24khz, 32khz, 44.1khz, 48khz, 96khz, 192khz with translation
using codec_resample.c
6. Various changes to RTP code required to properly handle the dynamic format list
and formats with attributes.
7. ConfBridge now dynamically jumps to the best possible sample rate. This allows
for conferences to take advantage of HD audio (Which sounds awesome)
8. Audiohooks are no longer limited to 8khz audio, and most effects have been
updated to take advantage of this such as Volume, DENOISE, PITCH_SHIFT.
9. codec_resample now uses its own code rather than depending on libresample.
-Organizational changes
Global format list is moved from frame.c to format.c
Various format specific functions moved from frame.c to format.c
Review: https://reviewboard.asterisk.org/r/1104/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308582 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-22 23:04:49 +00:00
static struct ast_frame * audiohook_list_translate_to_slin ( struct ast_audiohook_list * audiohook_list ,
enum ast_audiohook_direction direction , struct ast_frame * frame )
{
struct ast_audiohook_translate * in_translate = ( direction = = AST_AUDIOHOOK_DIRECTION_READ ?
& audiohook_list - > in_translate [ 0 ] : & audiohook_list - > in_translate [ 1 ] ) ;
struct ast_frame * new_frame = frame ;
struct ast_format tmp_fmt ;
enum ast_format_id slin_id ;
/* If we are capable of maintaining doing samplerates other that 8khz, update
* the internal audiohook_list ' s rate and higher samplerate audio arrives . By
* updating the list ' s rate , all the audiohooks in the list will be updated as well
* as the are written and read from . */
if ( audiohook_list - > native_slin_compatible ) {
audiohook_list - > list_internal_samp_rate =
MAX ( ast_format_rate ( & frame - > subclass . format ) , audiohook_list - > list_internal_samp_rate ) ;
}
slin_id = ast_format_slin_by_rate ( audiohook_list - > list_internal_samp_rate ) ;
if ( frame - > subclass . format . id = = slin_id ) {
return new_frame ;
}
if ( ast_format_cmp ( & frame - > subclass . format , & in_translate - > format ) = = AST_FORMAT_CMP_NOT_EQUAL ) {
if ( in_translate - > trans_pvt ) {
ast_translator_free_path ( in_translate - > trans_pvt ) ;
}
if ( ! ( in_translate - > trans_pvt = ast_translator_build_path ( ast_format_set ( & tmp_fmt , slin_id , 0 ) , & frame - > subclass . format ) ) ) {
return NULL ;
}
ast_format_copy ( & in_translate - > format , & frame - > subclass . format ) ;
}
if ( ! ( new_frame = ast_translate ( in_translate - > trans_pvt , frame , 0 ) ) ) {
return NULL ;
}
return new_frame ;
}
static struct ast_frame * audiohook_list_translate_to_native ( struct ast_audiohook_list * audiohook_list ,
enum ast_audiohook_direction direction , struct ast_frame * slin_frame , struct ast_format * outformat )
{
struct ast_audiohook_translate * out_translate = ( direction = = AST_AUDIOHOOK_DIRECTION_READ ? & audiohook_list - > out_translate [ 0 ] : & audiohook_list - > out_translate [ 1 ] ) ;
struct ast_frame * outframe = NULL ;
if ( ast_format_cmp ( & slin_frame - > subclass . format , outformat ) = = AST_FORMAT_CMP_NOT_EQUAL ) {
/* rebuild translators if necessary */
if ( ast_format_cmp ( & out_translate - > format , outformat ) = = AST_FORMAT_CMP_NOT_EQUAL ) {
if ( out_translate - > trans_pvt ) {
ast_translator_free_path ( out_translate - > trans_pvt ) ;
}
if ( ! ( out_translate - > trans_pvt = ast_translator_build_path ( outformat , & slin_frame - > subclass . format ) ) ) {
return NULL ;
}
ast_format_copy ( & out_translate - > format , outformat ) ;
}
/* translate back to the format the frame came in as. */
if ( ! ( outframe = ast_translate ( out_translate - > trans_pvt , slin_frame , 0 ) ) ) {
return NULL ;
}
}
return outframe ;
}
2010-04-29 15:33:27 +00:00
/*!
* \ brief Pass an AUDIO frame off to be handled by the audiohook core
*
* \ details
* This function has 3 ast_frames and 3 parts to handle each . At the beginning of this
* function all 3 frames , start_frame , middle_frame , and end_frame point to the initial
* input frame .
*
* Part_1 : Translate the start_frame into SLINEAR audio if it is not already in that
* format . The result of this part is middle_frame is guaranteed to be in
* SLINEAR format for Part_2 .
* Part_2 : Send middle_frame off to spies and manipulators . At this point middle_frame is
* either a new frame as result of the translation , or points directly to the start_frame
Media Project Phase2: SILK 8khz-24khz, SLINEAR 8khz-192khz, SPEEX 32khz, hd audio ConfBridge, and other stuff
-Functional changes
1. Dynamic global format list build by codecs defined in codecs.conf
2. SILK 8khz, 12khz, 16khz, and 24khz with custom attributes defined in codecs.conf
3. Negotiation of SILK attributes in chan_sip.
4. SPEEX 32khz with translation
5. SLINEAR 8khz, 12khz, 24khz, 32khz, 44.1khz, 48khz, 96khz, 192khz with translation
using codec_resample.c
6. Various changes to RTP code required to properly handle the dynamic format list
and formats with attributes.
7. ConfBridge now dynamically jumps to the best possible sample rate. This allows
for conferences to take advantage of HD audio (Which sounds awesome)
8. Audiohooks are no longer limited to 8khz audio, and most effects have been
updated to take advantage of this such as Volume, DENOISE, PITCH_SHIFT.
9. codec_resample now uses its own code rather than depending on libresample.
-Organizational changes
Global format list is moved from frame.c to format.c
Various format specific functions moved from frame.c to format.c
Review: https://reviewboard.asterisk.org/r/1104/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308582 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-22 23:04:49 +00:00
* because no translation to SLINEAR audio was required .
* Part_3 : Translate end_frame ' s audio back into the format of start frame if necessary . This
* is only necessary if manipulation of middle_frame occurred .
2010-04-29 15:33:27 +00:00
*
2007-08-08 19:30:52 +00:00
* \ param chan Channel that the list is coming off of
* \ param audiohook_list List of audiohooks
* \ param direction Direction frame is coming in from
* \ param frame The frame itself
* \ return Return frame on success , NULL on failure
*/
static struct ast_frame * audio_audiohook_write_list ( struct ast_channel * chan , struct ast_audiohook_list * audiohook_list , enum ast_audiohook_direction direction , struct ast_frame * frame )
{
struct ast_frame * start_frame = frame , * middle_frame = frame , * end_frame = frame ;
struct ast_audiohook * audiohook = NULL ;
Media Project Phase2: SILK 8khz-24khz, SLINEAR 8khz-192khz, SPEEX 32khz, hd audio ConfBridge, and other stuff
-Functional changes
1. Dynamic global format list build by codecs defined in codecs.conf
2. SILK 8khz, 12khz, 16khz, and 24khz with custom attributes defined in codecs.conf
3. Negotiation of SILK attributes in chan_sip.
4. SPEEX 32khz with translation
5. SLINEAR 8khz, 12khz, 24khz, 32khz, 44.1khz, 48khz, 96khz, 192khz with translation
using codec_resample.c
6. Various changes to RTP code required to properly handle the dynamic format list
and formats with attributes.
7. ConfBridge now dynamically jumps to the best possible sample rate. This allows
for conferences to take advantage of HD audio (Which sounds awesome)
8. Audiohooks are no longer limited to 8khz audio, and most effects have been
updated to take advantage of this such as Volume, DENOISE, PITCH_SHIFT.
9. codec_resample now uses its own code rather than depending on libresample.
-Organizational changes
Global format list is moved from frame.c to format.c
Various format specific functions moved from frame.c to format.c
Review: https://reviewboard.asterisk.org/r/1104/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308582 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-22 23:04:49 +00:00
int samples ;
int middle_frame_manipulated = 0 ;
int removed = 0 ;
2009-10-20 22:09:07 +00:00
2010-04-29 15:33:27 +00:00
/* ---Part_1. translate start_frame to SLINEAR if necessary. */
Media Project Phase2: SILK 8khz-24khz, SLINEAR 8khz-192khz, SPEEX 32khz, hd audio ConfBridge, and other stuff
-Functional changes
1. Dynamic global format list build by codecs defined in codecs.conf
2. SILK 8khz, 12khz, 16khz, and 24khz with custom attributes defined in codecs.conf
3. Negotiation of SILK attributes in chan_sip.
4. SPEEX 32khz with translation
5. SLINEAR 8khz, 12khz, 24khz, 32khz, 44.1khz, 48khz, 96khz, 192khz with translation
using codec_resample.c
6. Various changes to RTP code required to properly handle the dynamic format list
and formats with attributes.
7. ConfBridge now dynamically jumps to the best possible sample rate. This allows
for conferences to take advantage of HD audio (Which sounds awesome)
8. Audiohooks are no longer limited to 8khz audio, and most effects have been
updated to take advantage of this such as Volume, DENOISE, PITCH_SHIFT.
9. codec_resample now uses its own code rather than depending on libresample.
-Organizational changes
Global format list is moved from frame.c to format.c
Various format specific functions moved from frame.c to format.c
Review: https://reviewboard.asterisk.org/r/1104/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308582 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-22 23:04:49 +00:00
if ( ! ( middle_frame = audiohook_list_translate_to_slin ( audiohook_list , direction , start_frame ) ) ) {
return frame ;
2007-08-08 19:30:52 +00:00
}
Media Project Phase2: SILK 8khz-24khz, SLINEAR 8khz-192khz, SPEEX 32khz, hd audio ConfBridge, and other stuff
-Functional changes
1. Dynamic global format list build by codecs defined in codecs.conf
2. SILK 8khz, 12khz, 16khz, and 24khz with custom attributes defined in codecs.conf
3. Negotiation of SILK attributes in chan_sip.
4. SPEEX 32khz with translation
5. SLINEAR 8khz, 12khz, 24khz, 32khz, 44.1khz, 48khz, 96khz, 192khz with translation
using codec_resample.c
6. Various changes to RTP code required to properly handle the dynamic format list
and formats with attributes.
7. ConfBridge now dynamically jumps to the best possible sample rate. This allows
for conferences to take advantage of HD audio (Which sounds awesome)
8. Audiohooks are no longer limited to 8khz audio, and most effects have been
updated to take advantage of this such as Volume, DENOISE, PITCH_SHIFT.
9. codec_resample now uses its own code rather than depending on libresample.
-Organizational changes
Global format list is moved from frame.c to format.c
Various format specific functions moved from frame.c to format.c
Review: https://reviewboard.asterisk.org/r/1104/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308582 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-22 23:04:49 +00:00
samples = middle_frame - > samples ;
2007-08-08 19:30:52 +00:00
2010-04-29 15:33:27 +00:00
/* ---Part_2: Send middle_frame to spy and manipulator lists. middle_frame is guaranteed to be SLINEAR here.*/
2007-08-08 19:30:52 +00:00
/* Queue up signed linear frame to each spy */
AST_LIST_TRAVERSE_SAFE_BEGIN ( & audiohook_list - > spy_list , audiohook , list ) {
ast_audiohook_lock ( audiohook ) ;
if ( audiohook - > status ! = AST_AUDIOHOOK_STATUS_RUNNING ) {
2007-11-08 05:28:47 +00:00
AST_LIST_REMOVE_CURRENT ( list ) ;
Media Project Phase2: SILK 8khz-24khz, SLINEAR 8khz-192khz, SPEEX 32khz, hd audio ConfBridge, and other stuff
-Functional changes
1. Dynamic global format list build by codecs defined in codecs.conf
2. SILK 8khz, 12khz, 16khz, and 24khz with custom attributes defined in codecs.conf
3. Negotiation of SILK attributes in chan_sip.
4. SPEEX 32khz with translation
5. SLINEAR 8khz, 12khz, 24khz, 32khz, 44.1khz, 48khz, 96khz, 192khz with translation
using codec_resample.c
6. Various changes to RTP code required to properly handle the dynamic format list
and formats with attributes.
7. ConfBridge now dynamically jumps to the best possible sample rate. This allows
for conferences to take advantage of HD audio (Which sounds awesome)
8. Audiohooks are no longer limited to 8khz audio, and most effects have been
updated to take advantage of this such as Volume, DENOISE, PITCH_SHIFT.
9. codec_resample now uses its own code rather than depending on libresample.
-Organizational changes
Global format list is moved from frame.c to format.c
Various format specific functions moved from frame.c to format.c
Review: https://reviewboard.asterisk.org/r/1104/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308582 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-22 23:04:49 +00:00
removed = 1 ;
2009-11-20 17:26:20 +00:00
ast_audiohook_update_status ( audiohook , AST_AUDIOHOOK_STATUS_DONE ) ;
2007-08-08 19:30:52 +00:00
ast_audiohook_unlock ( audiohook ) ;
continue ;
}
Media Project Phase2: SILK 8khz-24khz, SLINEAR 8khz-192khz, SPEEX 32khz, hd audio ConfBridge, and other stuff
-Functional changes
1. Dynamic global format list build by codecs defined in codecs.conf
2. SILK 8khz, 12khz, 16khz, and 24khz with custom attributes defined in codecs.conf
3. Negotiation of SILK attributes in chan_sip.
4. SPEEX 32khz with translation
5. SLINEAR 8khz, 12khz, 24khz, 32khz, 44.1khz, 48khz, 96khz, 192khz with translation
using codec_resample.c
6. Various changes to RTP code required to properly handle the dynamic format list
and formats with attributes.
7. ConfBridge now dynamically jumps to the best possible sample rate. This allows
for conferences to take advantage of HD audio (Which sounds awesome)
8. Audiohooks are no longer limited to 8khz audio, and most effects have been
updated to take advantage of this such as Volume, DENOISE, PITCH_SHIFT.
9. codec_resample now uses its own code rather than depending on libresample.
-Organizational changes
Global format list is moved from frame.c to format.c
Various format specific functions moved from frame.c to format.c
Review: https://reviewboard.asterisk.org/r/1104/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308582 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-22 23:04:49 +00:00
audiohook_set_internal_rate ( audiohook , audiohook_list - > list_internal_samp_rate , 1 ) ;
2007-08-08 19:30:52 +00:00
ast_audiohook_write_frame ( audiohook , direction , middle_frame ) ;
ast_audiohook_unlock ( audiohook ) ;
}
2009-04-10 03:55:27 +00:00
AST_LIST_TRAVERSE_SAFE_END ;
2007-08-08 19:30:52 +00:00
/* If this frame is being written out to the channel then we need to use whisper sources */
if ( direction = = AST_AUDIOHOOK_DIRECTION_WRITE & & ! AST_LIST_EMPTY ( & audiohook_list - > whisper_list ) ) {
int i = 0 ;
short read_buf [ samples ] , combine_buf [ samples ] , * data1 = NULL , * data2 = NULL ;
memset ( & combine_buf , 0 , sizeof ( combine_buf ) ) ;
AST_LIST_TRAVERSE_SAFE_BEGIN ( & audiohook_list - > whisper_list , audiohook , list ) {
ast_audiohook_lock ( audiohook ) ;
if ( audiohook - > status ! = AST_AUDIOHOOK_STATUS_RUNNING ) {
2007-11-08 05:28:47 +00:00
AST_LIST_REMOVE_CURRENT ( list ) ;
Media Project Phase2: SILK 8khz-24khz, SLINEAR 8khz-192khz, SPEEX 32khz, hd audio ConfBridge, and other stuff
-Functional changes
1. Dynamic global format list build by codecs defined in codecs.conf
2. SILK 8khz, 12khz, 16khz, and 24khz with custom attributes defined in codecs.conf
3. Negotiation of SILK attributes in chan_sip.
4. SPEEX 32khz with translation
5. SLINEAR 8khz, 12khz, 24khz, 32khz, 44.1khz, 48khz, 96khz, 192khz with translation
using codec_resample.c
6. Various changes to RTP code required to properly handle the dynamic format list
and formats with attributes.
7. ConfBridge now dynamically jumps to the best possible sample rate. This allows
for conferences to take advantage of HD audio (Which sounds awesome)
8. Audiohooks are no longer limited to 8khz audio, and most effects have been
updated to take advantage of this such as Volume, DENOISE, PITCH_SHIFT.
9. codec_resample now uses its own code rather than depending on libresample.
-Organizational changes
Global format list is moved from frame.c to format.c
Various format specific functions moved from frame.c to format.c
Review: https://reviewboard.asterisk.org/r/1104/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308582 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-22 23:04:49 +00:00
removed = 1 ;
2009-11-20 17:26:20 +00:00
ast_audiohook_update_status ( audiohook , AST_AUDIOHOOK_STATUS_DONE ) ;
2007-08-08 19:30:52 +00:00
ast_audiohook_unlock ( audiohook ) ;
continue ;
}
Media Project Phase2: SILK 8khz-24khz, SLINEAR 8khz-192khz, SPEEX 32khz, hd audio ConfBridge, and other stuff
-Functional changes
1. Dynamic global format list build by codecs defined in codecs.conf
2. SILK 8khz, 12khz, 16khz, and 24khz with custom attributes defined in codecs.conf
3. Negotiation of SILK attributes in chan_sip.
4. SPEEX 32khz with translation
5. SLINEAR 8khz, 12khz, 24khz, 32khz, 44.1khz, 48khz, 96khz, 192khz with translation
using codec_resample.c
6. Various changes to RTP code required to properly handle the dynamic format list
and formats with attributes.
7. ConfBridge now dynamically jumps to the best possible sample rate. This allows
for conferences to take advantage of HD audio (Which sounds awesome)
8. Audiohooks are no longer limited to 8khz audio, and most effects have been
updated to take advantage of this such as Volume, DENOISE, PITCH_SHIFT.
9. codec_resample now uses its own code rather than depending on libresample.
-Organizational changes
Global format list is moved from frame.c to format.c
Various format specific functions moved from frame.c to format.c
Review: https://reviewboard.asterisk.org/r/1104/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308582 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-22 23:04:49 +00:00
audiohook_set_internal_rate ( audiohook , audiohook_list - > list_internal_samp_rate , 1 ) ;
2007-08-08 19:30:52 +00:00
if ( ast_slinfactory_available ( & audiohook - > write_factory ) > = samples & & ast_slinfactory_read ( & audiohook - > write_factory , read_buf , samples ) ) {
/* Take audio from this whisper source and combine it into our main buffer */
for ( i = 0 , data1 = combine_buf , data2 = read_buf ; i < samples ; i + + , data1 + + , data2 + + )
ast_slinear_saturated_add ( data1 , data2 ) ;
}
ast_audiohook_unlock ( audiohook ) ;
}
2009-04-10 03:55:27 +00:00
AST_LIST_TRAVERSE_SAFE_END ;
2007-08-08 19:30:52 +00:00
/* We take all of the combined whisper sources and combine them into the audio being written out */
Media Project Phase2: SILK 8khz-24khz, SLINEAR 8khz-192khz, SPEEX 32khz, hd audio ConfBridge, and other stuff
-Functional changes
1. Dynamic global format list build by codecs defined in codecs.conf
2. SILK 8khz, 12khz, 16khz, and 24khz with custom attributes defined in codecs.conf
3. Negotiation of SILK attributes in chan_sip.
4. SPEEX 32khz with translation
5. SLINEAR 8khz, 12khz, 24khz, 32khz, 44.1khz, 48khz, 96khz, 192khz with translation
using codec_resample.c
6. Various changes to RTP code required to properly handle the dynamic format list
and formats with attributes.
7. ConfBridge now dynamically jumps to the best possible sample rate. This allows
for conferences to take advantage of HD audio (Which sounds awesome)
8. Audiohooks are no longer limited to 8khz audio, and most effects have been
updated to take advantage of this such as Volume, DENOISE, PITCH_SHIFT.
9. codec_resample now uses its own code rather than depending on libresample.
-Organizational changes
Global format list is moved from frame.c to format.c
Various format specific functions moved from frame.c to format.c
Review: https://reviewboard.asterisk.org/r/1104/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308582 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-22 23:04:49 +00:00
for ( i = 0 , data1 = middle_frame - > data . ptr , data2 = combine_buf ; i < samples ; i + + , data1 + + , data2 + + ) {
2007-08-08 19:30:52 +00:00
ast_slinear_saturated_add ( data1 , data2 ) ;
Media Project Phase2: SILK 8khz-24khz, SLINEAR 8khz-192khz, SPEEX 32khz, hd audio ConfBridge, and other stuff
-Functional changes
1. Dynamic global format list build by codecs defined in codecs.conf
2. SILK 8khz, 12khz, 16khz, and 24khz with custom attributes defined in codecs.conf
3. Negotiation of SILK attributes in chan_sip.
4. SPEEX 32khz with translation
5. SLINEAR 8khz, 12khz, 24khz, 32khz, 44.1khz, 48khz, 96khz, 192khz with translation
using codec_resample.c
6. Various changes to RTP code required to properly handle the dynamic format list
and formats with attributes.
7. ConfBridge now dynamically jumps to the best possible sample rate. This allows
for conferences to take advantage of HD audio (Which sounds awesome)
8. Audiohooks are no longer limited to 8khz audio, and most effects have been
updated to take advantage of this such as Volume, DENOISE, PITCH_SHIFT.
9. codec_resample now uses its own code rather than depending on libresample.
-Organizational changes
Global format list is moved from frame.c to format.c
Various format specific functions moved from frame.c to format.c
Review: https://reviewboard.asterisk.org/r/1104/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308582 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-22 23:04:49 +00:00
}
middle_frame_manipulated = 1 ;
2007-08-08 19:30:52 +00:00
}
/* Pass off frame to manipulate audiohooks */
if ( ! AST_LIST_EMPTY ( & audiohook_list - > manipulate_list ) ) {
AST_LIST_TRAVERSE_SAFE_BEGIN ( & audiohook_list - > manipulate_list , audiohook , list ) {
ast_audiohook_lock ( audiohook ) ;
if ( audiohook - > status ! = AST_AUDIOHOOK_STATUS_RUNNING ) {
2007-11-08 05:28:47 +00:00
AST_LIST_REMOVE_CURRENT ( list ) ;
Media Project Phase2: SILK 8khz-24khz, SLINEAR 8khz-192khz, SPEEX 32khz, hd audio ConfBridge, and other stuff
-Functional changes
1. Dynamic global format list build by codecs defined in codecs.conf
2. SILK 8khz, 12khz, 16khz, and 24khz with custom attributes defined in codecs.conf
3. Negotiation of SILK attributes in chan_sip.
4. SPEEX 32khz with translation
5. SLINEAR 8khz, 12khz, 24khz, 32khz, 44.1khz, 48khz, 96khz, 192khz with translation
using codec_resample.c
6. Various changes to RTP code required to properly handle the dynamic format list
and formats with attributes.
7. ConfBridge now dynamically jumps to the best possible sample rate. This allows
for conferences to take advantage of HD audio (Which sounds awesome)
8. Audiohooks are no longer limited to 8khz audio, and most effects have been
updated to take advantage of this such as Volume, DENOISE, PITCH_SHIFT.
9. codec_resample now uses its own code rather than depending on libresample.
-Organizational changes
Global format list is moved from frame.c to format.c
Various format specific functions moved from frame.c to format.c
Review: https://reviewboard.asterisk.org/r/1104/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308582 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-22 23:04:49 +00:00
removed = 1 ;
2009-11-20 17:26:20 +00:00
ast_audiohook_update_status ( audiohook , AST_AUDIOHOOK_STATUS_DONE ) ;
2007-08-08 19:30:52 +00:00
ast_audiohook_unlock ( audiohook ) ;
/* We basically drop all of our links to the manipulate audiohook and prod it to do it's own destructive things */
audiohook - > manipulate_callback ( audiohook , chan , NULL , direction ) ;
continue ;
}
Media Project Phase2: SILK 8khz-24khz, SLINEAR 8khz-192khz, SPEEX 32khz, hd audio ConfBridge, and other stuff
-Functional changes
1. Dynamic global format list build by codecs defined in codecs.conf
2. SILK 8khz, 12khz, 16khz, and 24khz with custom attributes defined in codecs.conf
3. Negotiation of SILK attributes in chan_sip.
4. SPEEX 32khz with translation
5. SLINEAR 8khz, 12khz, 24khz, 32khz, 44.1khz, 48khz, 96khz, 192khz with translation
using codec_resample.c
6. Various changes to RTP code required to properly handle the dynamic format list
and formats with attributes.
7. ConfBridge now dynamically jumps to the best possible sample rate. This allows
for conferences to take advantage of HD audio (Which sounds awesome)
8. Audiohooks are no longer limited to 8khz audio, and most effects have been
updated to take advantage of this such as Volume, DENOISE, PITCH_SHIFT.
9. codec_resample now uses its own code rather than depending on libresample.
-Organizational changes
Global format list is moved from frame.c to format.c
Various format specific functions moved from frame.c to format.c
Review: https://reviewboard.asterisk.org/r/1104/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308582 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-22 23:04:49 +00:00
audiohook_set_internal_rate ( audiohook , audiohook_list - > list_internal_samp_rate , 1 ) ;
2010-04-29 15:33:27 +00:00
/* Feed in frame to manipulation. */
2009-10-20 22:09:07 +00:00
if ( audiohook - > manipulate_callback ( audiohook , chan , middle_frame , direction ) ) {
2010-04-29 15:33:27 +00:00
/* XXX IGNORE FAILURE */
/* If the manipulation fails then the frame will be returned in its original state.
* Since there are potentially more manipulator callbacks in the list , no action should
* be taken here to exit early . */
2009-10-20 22:09:07 +00:00
}
2007-08-08 19:30:52 +00:00
ast_audiohook_unlock ( audiohook ) ;
}
2009-04-10 03:55:27 +00:00
AST_LIST_TRAVERSE_SAFE_END ;
Media Project Phase2: SILK 8khz-24khz, SLINEAR 8khz-192khz, SPEEX 32khz, hd audio ConfBridge, and other stuff
-Functional changes
1. Dynamic global format list build by codecs defined in codecs.conf
2. SILK 8khz, 12khz, 16khz, and 24khz with custom attributes defined in codecs.conf
3. Negotiation of SILK attributes in chan_sip.
4. SPEEX 32khz with translation
5. SLINEAR 8khz, 12khz, 24khz, 32khz, 44.1khz, 48khz, 96khz, 192khz with translation
using codec_resample.c
6. Various changes to RTP code required to properly handle the dynamic format list
and formats with attributes.
7. ConfBridge now dynamically jumps to the best possible sample rate. This allows
for conferences to take advantage of HD audio (Which sounds awesome)
8. Audiohooks are no longer limited to 8khz audio, and most effects have been
updated to take advantage of this such as Volume, DENOISE, PITCH_SHIFT.
9. codec_resample now uses its own code rather than depending on libresample.
-Organizational changes
Global format list is moved from frame.c to format.c
Various format specific functions moved from frame.c to format.c
Review: https://reviewboard.asterisk.org/r/1104/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308582 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-22 23:04:49 +00:00
middle_frame_manipulated = 1 ;
2007-08-08 19:30:52 +00:00
}
2010-04-29 15:33:27 +00:00
/* ---Part_3: Decide what to do with the end_frame (whether to transcode or not) */
Media Project Phase2: SILK 8khz-24khz, SLINEAR 8khz-192khz, SPEEX 32khz, hd audio ConfBridge, and other stuff
-Functional changes
1. Dynamic global format list build by codecs defined in codecs.conf
2. SILK 8khz, 12khz, 16khz, and 24khz with custom attributes defined in codecs.conf
3. Negotiation of SILK attributes in chan_sip.
4. SPEEX 32khz with translation
5. SLINEAR 8khz, 12khz, 24khz, 32khz, 44.1khz, 48khz, 96khz, 192khz with translation
using codec_resample.c
6. Various changes to RTP code required to properly handle the dynamic format list
and formats with attributes.
7. ConfBridge now dynamically jumps to the best possible sample rate. This allows
for conferences to take advantage of HD audio (Which sounds awesome)
8. Audiohooks are no longer limited to 8khz audio, and most effects have been
updated to take advantage of this such as Volume, DENOISE, PITCH_SHIFT.
9. codec_resample now uses its own code rather than depending on libresample.
-Organizational changes
Global format list is moved from frame.c to format.c
Various format specific functions moved from frame.c to format.c
Review: https://reviewboard.asterisk.org/r/1104/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308582 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-22 23:04:49 +00:00
if ( middle_frame_manipulated ) {
if ( ! ( end_frame = audiohook_list_translate_to_native ( audiohook_list , direction , middle_frame , & start_frame - > subclass . format ) ) ) {
/* translation failed, so just pass back the input frame */
end_frame = start_frame ;
2007-08-08 19:30:52 +00:00
}
} else {
Media Project Phase2: SILK 8khz-24khz, SLINEAR 8khz-192khz, SPEEX 32khz, hd audio ConfBridge, and other stuff
-Functional changes
1. Dynamic global format list build by codecs defined in codecs.conf
2. SILK 8khz, 12khz, 16khz, and 24khz with custom attributes defined in codecs.conf
3. Negotiation of SILK attributes in chan_sip.
4. SPEEX 32khz with translation
5. SLINEAR 8khz, 12khz, 24khz, 32khz, 44.1khz, 48khz, 96khz, 192khz with translation
using codec_resample.c
6. Various changes to RTP code required to properly handle the dynamic format list
and formats with attributes.
7. ConfBridge now dynamically jumps to the best possible sample rate. This allows
for conferences to take advantage of HD audio (Which sounds awesome)
8. Audiohooks are no longer limited to 8khz audio, and most effects have been
updated to take advantage of this such as Volume, DENOISE, PITCH_SHIFT.
9. codec_resample now uses its own code rather than depending on libresample.
-Organizational changes
Global format list is moved from frame.c to format.c
Various format specific functions moved from frame.c to format.c
Review: https://reviewboard.asterisk.org/r/1104/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308582 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-22 23:04:49 +00:00
end_frame = start_frame ;
}
/* clean up our middle_frame if required */
if ( middle_frame ! = end_frame ) {
2010-04-29 15:33:27 +00:00
ast_frfree ( middle_frame ) ;
Media Project Phase2: SILK 8khz-24khz, SLINEAR 8khz-192khz, SPEEX 32khz, hd audio ConfBridge, and other stuff
-Functional changes
1. Dynamic global format list build by codecs defined in codecs.conf
2. SILK 8khz, 12khz, 16khz, and 24khz with custom attributes defined in codecs.conf
3. Negotiation of SILK attributes in chan_sip.
4. SPEEX 32khz with translation
5. SLINEAR 8khz, 12khz, 24khz, 32khz, 44.1khz, 48khz, 96khz, 192khz with translation
using codec_resample.c
6. Various changes to RTP code required to properly handle the dynamic format list
and formats with attributes.
7. ConfBridge now dynamically jumps to the best possible sample rate. This allows
for conferences to take advantage of HD audio (Which sounds awesome)
8. Audiohooks are no longer limited to 8khz audio, and most effects have been
updated to take advantage of this such as Volume, DENOISE, PITCH_SHIFT.
9. codec_resample now uses its own code rather than depending on libresample.
-Organizational changes
Global format list is moved from frame.c to format.c
Various format specific functions moved from frame.c to format.c
Review: https://reviewboard.asterisk.org/r/1104/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308582 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-22 23:04:49 +00:00
middle_frame = NULL ;
}
/* Before returning, if an audiohook got removed, reset samplerate compatibility */
if ( removed ) {
audiohook_list_set_samplerate_compatibility ( audiohook_list ) ;
2007-08-08 19:30:52 +00:00
}
return end_frame ;
}
2010-07-27 20:59:16 +00:00
int ast_audiohook_write_list_empty ( struct ast_audiohook_list * audiohook_list )
{
if ( AST_LIST_EMPTY ( & audiohook_list - > spy_list ) & &
AST_LIST_EMPTY ( & audiohook_list - > whisper_list ) & &
AST_LIST_EMPTY ( & audiohook_list - > manipulate_list ) ) {
return 1 ;
}
return 0 ;
}
2007-08-08 19:30:52 +00:00
/*! \brief Pass a frame off to be handled by the audiohook core
* \ param chan Channel that the list is coming off of
* \ param audiohook_list List of audiohooks
* \ param direction Direction frame is coming in from
* \ param frame The frame itself
* \ return Return frame on success , NULL on failure
*/
struct ast_frame * ast_audiohook_write_list ( struct ast_channel * chan , struct ast_audiohook_list * audiohook_list , enum ast_audiohook_direction direction , struct ast_frame * frame )
{
/* Pass off frame to it's respective list write function */
if ( frame - > frametype = = AST_FRAME_VOICE )
return audio_audiohook_write_list ( chan , audiohook_list , direction , frame ) ;
else if ( frame - > frametype = = AST_FRAME_DTMF )
return dtmf_audiohook_write_list ( chan , audiohook_list , direction , frame ) ;
else
return frame ;
}
/*! \brief Wait for audiohook trigger to be triggered
* \ param audiohook Audiohook to wait on
*/
void ast_audiohook_trigger_wait ( struct ast_audiohook * audiohook )
{
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struct timeval wait ;
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struct timespec ts ;
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wait = ast_tvadd ( ast_tvnow ( ) , ast_samp2tv ( 50000 , 1000 ) ) ;
ts . tv_sec = wait . tv_sec ;
ts . tv_nsec = wait . tv_usec * 1000 ;
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ast_cond_timedwait ( & audiohook - > trigger , & audiohook - > lock , & ts ) ;
return ;
}
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/* Count number of channel audiohooks by type, regardless of type */
int ast_channel_audiohook_count_by_source ( struct ast_channel * chan , const char * source , enum ast_audiohook_type type )
{
int count = 0 ;
struct ast_audiohook * ah = NULL ;
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if ( ! ast_channel_audiohooks ( chan ) )
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return - 1 ;
switch ( type ) {
case AST_AUDIOHOOK_TYPE_SPY :
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AST_LIST_TRAVERSE ( & ast_channel_audiohooks ( chan ) - > spy_list , ah , list ) {
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if ( ! strcmp ( ah - > source , source ) ) {
count + + ;
}
}
break ;
case AST_AUDIOHOOK_TYPE_WHISPER :
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AST_LIST_TRAVERSE ( & ast_channel_audiohooks ( chan ) - > whisper_list , ah , list ) {
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if ( ! strcmp ( ah - > source , source ) ) {
count + + ;
}
}
break ;
case AST_AUDIOHOOK_TYPE_MANIPULATE :
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AST_LIST_TRAVERSE ( & ast_channel_audiohooks ( chan ) - > manipulate_list , ah , list ) {
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if ( ! strcmp ( ah - > source , source ) ) {
count + + ;
}
}
break ;
default :
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ast_debug ( 1 , " Invalid audiohook type supplied, (%d) \n " , type ) ;
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return - 1 ;
}
return count ;
}
/* Count number of channel audiohooks by type that are running */
int ast_channel_audiohook_count_by_source_running ( struct ast_channel * chan , const char * source , enum ast_audiohook_type type )
{
int count = 0 ;
struct ast_audiohook * ah = NULL ;
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if ( ! ast_channel_audiohooks ( chan ) )
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return - 1 ;
switch ( type ) {
case AST_AUDIOHOOK_TYPE_SPY :
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AST_LIST_TRAVERSE ( & ast_channel_audiohooks ( chan ) - > spy_list , ah , list ) {
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if ( ( ! strcmp ( ah - > source , source ) ) & & ( ah - > status = = AST_AUDIOHOOK_STATUS_RUNNING ) )
count + + ;
}
break ;
case AST_AUDIOHOOK_TYPE_WHISPER :
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AST_LIST_TRAVERSE ( & ast_channel_audiohooks ( chan ) - > whisper_list , ah , list ) {
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if ( ( ! strcmp ( ah - > source , source ) ) & & ( ah - > status = = AST_AUDIOHOOK_STATUS_RUNNING ) )
count + + ;
}
break ;
case AST_AUDIOHOOK_TYPE_MANIPULATE :
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AST_LIST_TRAVERSE ( & ast_channel_audiohooks ( chan ) - > manipulate_list , ah , list ) {
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if ( ( ! strcmp ( ah - > source , source ) ) & & ( ah - > status = = AST_AUDIOHOOK_STATUS_RUNNING ) )
count + + ;
}
break ;
default :
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ast_debug ( 1 , " Invalid audiohook type supplied, (%d) \n " , type ) ;
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return - 1 ;
}
return count ;
}
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/*! \brief Audiohook volume adjustment structure */
struct audiohook_volume {
struct ast_audiohook audiohook ; /*!< Audiohook attached to the channel */
int read_adjustment ; /*!< Value to adjust frames read from the channel by */
int write_adjustment ; /*!< Value to adjust frames written to the channel by */
} ;
/*! \brief Callback used to destroy the audiohook volume datastore
* \ param data Volume information structure
* \ return Returns nothing
*/
static void audiohook_volume_destroy ( void * data )
{
struct audiohook_volume * audiohook_volume = data ;
/* Destroy the audiohook as it is no longer in use */
ast_audiohook_destroy ( & audiohook_volume - > audiohook ) ;
/* Finally free ourselves, we are of no more use */
ast_free ( audiohook_volume ) ;
return ;
}
/*! \brief Datastore used to store audiohook volume information */
static const struct ast_datastore_info audiohook_volume_datastore = {
. type = " Volume " ,
. destroy = audiohook_volume_destroy ,
} ;
/*! \brief Helper function which actually gets called by audiohooks to perform the adjustment
* \ param audiohook Audiohook attached to the channel
* \ param chan Channel we are attached to
* \ param frame Frame of audio we want to manipulate
* \ param direction Direction the audio came in from
* \ return Returns 0 on success , - 1 on failure
*/
static int audiohook_volume_callback ( struct ast_audiohook * audiohook , struct ast_channel * chan , struct ast_frame * frame , enum ast_audiohook_direction direction )
{
struct ast_datastore * datastore = NULL ;
struct audiohook_volume * audiohook_volume = NULL ;
int * gain = NULL ;
/* If the audiohook is shutting down don't even bother */
if ( audiohook - > status = = AST_AUDIOHOOK_STATUS_DONE ) {
return 0 ;
}
/* Try to find the datastore containg adjustment information, if we can't just bail out */
if ( ! ( datastore = ast_channel_datastore_find ( chan , & audiohook_volume_datastore , NULL ) ) ) {
return 0 ;
}
audiohook_volume = datastore - > data ;
/* Based on direction grab the appropriate adjustment value */
if ( direction = = AST_AUDIOHOOK_DIRECTION_READ ) {
gain = & audiohook_volume - > read_adjustment ;
} else if ( direction = = AST_AUDIOHOOK_DIRECTION_WRITE ) {
gain = & audiohook_volume - > write_adjustment ;
}
/* If an adjustment value is present modify the frame */
if ( gain & & * gain ) {
ast_frame_adjust_volume ( frame , * gain ) ;
}
return 0 ;
}
/*! \brief Helper function which finds and optionally creates an audiohook_volume_datastore datastore on a channel
* \ param chan Channel to look on
* \ param create Whether to create the datastore if not found
* \ return Returns audiohook_volume structure on success , NULL on failure
*/
static struct audiohook_volume * audiohook_volume_get ( struct ast_channel * chan , int create )
{
struct ast_datastore * datastore = NULL ;
struct audiohook_volume * audiohook_volume = NULL ;
/* If we are able to find the datastore return the contents (which is actually an audiohook_volume structure) */
if ( ( datastore = ast_channel_datastore_find ( chan , & audiohook_volume_datastore , NULL ) ) ) {
return datastore - > data ;
}
/* If we are not allowed to create a datastore or if we fail to create a datastore, bail out now as we have nothing for them */
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if ( ! create | | ! ( datastore = ast_datastore_alloc ( & audiohook_volume_datastore , NULL ) ) ) {
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return NULL ;
}
/* Create a new audiohook_volume structure to contain our adjustments and audiohook */
if ( ! ( audiohook_volume = ast_calloc ( 1 , sizeof ( * audiohook_volume ) ) ) ) {
2008-08-05 16:56:11 +00:00
ast_datastore_free ( datastore ) ;
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return NULL ;
}
/* Setup our audiohook structure so we can manipulate the audio */
Media Project Phase2: SILK 8khz-24khz, SLINEAR 8khz-192khz, SPEEX 32khz, hd audio ConfBridge, and other stuff
-Functional changes
1. Dynamic global format list build by codecs defined in codecs.conf
2. SILK 8khz, 12khz, 16khz, and 24khz with custom attributes defined in codecs.conf
3. Negotiation of SILK attributes in chan_sip.
4. SPEEX 32khz with translation
5. SLINEAR 8khz, 12khz, 24khz, 32khz, 44.1khz, 48khz, 96khz, 192khz with translation
using codec_resample.c
6. Various changes to RTP code required to properly handle the dynamic format list
and formats with attributes.
7. ConfBridge now dynamically jumps to the best possible sample rate. This allows
for conferences to take advantage of HD audio (Which sounds awesome)
8. Audiohooks are no longer limited to 8khz audio, and most effects have been
updated to take advantage of this such as Volume, DENOISE, PITCH_SHIFT.
9. codec_resample now uses its own code rather than depending on libresample.
-Organizational changes
Global format list is moved from frame.c to format.c
Various format specific functions moved from frame.c to format.c
Review: https://reviewboard.asterisk.org/r/1104/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308582 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-22 23:04:49 +00:00
ast_audiohook_init ( & audiohook_volume - > audiohook , AST_AUDIOHOOK_TYPE_MANIPULATE , " Volume " , AST_AUDIOHOOK_MANIPULATE_ALL_RATES ) ;
2008-03-21 17:58:59 +00:00
audiohook_volume - > audiohook . manipulate_callback = audiohook_volume_callback ;
/* Attach the audiohook_volume blob to the datastore and attach to the channel */
datastore - > data = audiohook_volume ;
ast_channel_datastore_add ( chan , datastore ) ;
/* All is well... put the audiohook into motion */
ast_audiohook_attach ( chan , & audiohook_volume - > audiohook ) ;
return audiohook_volume ;
}
/*! \brief Adjust the volume on frames read from or written to a channel
* \ param chan Channel to muck with
* \ param direction Direction to set on
* \ param volume Value to adjust the volume by
* \ return Returns 0 on success , - 1 on failure
*/
int ast_audiohook_volume_set ( struct ast_channel * chan , enum ast_audiohook_direction direction , int volume )
{
struct audiohook_volume * audiohook_volume = NULL ;
/* Attempt to find the audiohook volume information, but only create it if we are not setting the adjustment value to zero */
if ( ! ( audiohook_volume = audiohook_volume_get ( chan , ( volume ? 1 : 0 ) ) ) ) {
return - 1 ;
}
/* Now based on the direction set the proper value */
if ( direction = = AST_AUDIOHOOK_DIRECTION_READ | | direction = = AST_AUDIOHOOK_DIRECTION_BOTH ) {
audiohook_volume - > read_adjustment = volume ;
2009-03-02 14:13:45 +00:00
}
if ( direction = = AST_AUDIOHOOK_DIRECTION_WRITE | | direction = = AST_AUDIOHOOK_DIRECTION_BOTH ) {
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audiohook_volume - > write_adjustment = volume ;
}
return 0 ;
}
/*! \brief Retrieve the volume adjustment value on frames read from or written to a channel
* \ param chan Channel to retrieve volume adjustment from
* \ param direction Direction to retrieve
* \ return Returns adjustment value
*/
int ast_audiohook_volume_get ( struct ast_channel * chan , enum ast_audiohook_direction direction )
{
struct audiohook_volume * audiohook_volume = NULL ;
int adjustment = 0 ;
/* Attempt to find the audiohook volume information, but do not create it as we only want to look at the values */
if ( ! ( audiohook_volume = audiohook_volume_get ( chan , 0 ) ) ) {
return 0 ;
}
/* Grab the adjustment value based on direction given */
if ( direction = = AST_AUDIOHOOK_DIRECTION_READ ) {
adjustment = audiohook_volume - > read_adjustment ;
} else if ( direction = = AST_AUDIOHOOK_DIRECTION_WRITE ) {
adjustment = audiohook_volume - > write_adjustment ;
}
return adjustment ;
}
/*! \brief Adjust the volume on frames read from or written to a channel
* \ param chan Channel to muck with
* \ param direction Direction to increase
* \ param volume Value to adjust the adjustment by
* \ return Returns 0 on success , - 1 on failure
*/
int ast_audiohook_volume_adjust ( struct ast_channel * chan , enum ast_audiohook_direction direction , int volume )
{
struct audiohook_volume * audiohook_volume = NULL ;
/* Attempt to find the audiohook volume information, and create an audiohook if none exists */
if ( ! ( audiohook_volume = audiohook_volume_get ( chan , 1 ) ) ) {
return - 1 ;
}
/* Based on the direction change the specific adjustment value */
if ( direction = = AST_AUDIOHOOK_DIRECTION_READ | | direction = = AST_AUDIOHOOK_DIRECTION_BOTH ) {
audiohook_volume - > read_adjustment + = volume ;
2009-03-02 14:13:45 +00:00
}
if ( direction = = AST_AUDIOHOOK_DIRECTION_WRITE | | direction = = AST_AUDIOHOOK_DIRECTION_BOTH ) {
2008-03-21 17:58:59 +00:00
audiohook_volume - > write_adjustment + = volume ;
}
return 0 ;
}
2010-04-21 11:27:27 +00:00
/*! \brief Mute frames read from or written to a channel
* \ param chan Channel to muck with
* \ param source Type of audiohook
* \ param flag which flag to set / clear
* \ param clear set or clear
* \ return Returns 0 on success , - 1 on failure
*/
int ast_audiohook_set_mute ( struct ast_channel * chan , const char * source , enum ast_audiohook_flags flag , int clear )
{
struct ast_audiohook * audiohook = NULL ;
ast_channel_lock ( chan ) ;
/* Ensure the channel has audiohooks on it */
2012-02-20 23:43:27 +00:00
if ( ! ast_channel_audiohooks ( chan ) ) {
2010-04-21 11:27:27 +00:00
ast_channel_unlock ( chan ) ;
return - 1 ;
}
2012-02-20 23:43:27 +00:00
audiohook = find_audiohook_by_source ( ast_channel_audiohooks ( chan ) , source ) ;
2010-04-21 11:27:27 +00:00
if ( audiohook ) {
if ( clear ) {
ast_clear_flag ( audiohook , flag ) ;
} else {
ast_set_flag ( audiohook , flag ) ;
}
}
ast_channel_unlock ( chan ) ;
return ( audiohook ? 0 : - 1 ) ;
}