[Asterisk-code-review] chan_oss: Remove deprecated module. (asterisk[master])
George Joseph
asteriskteam at digium.com
Wed Aug 18 11:12:11 CDT 2021
George Joseph has submitted this change. ( https://gerrit.asterisk.org/c/asterisk/+/16311 )
Change subject: chan_oss: Remove deprecated module.
......................................................................
chan_oss: Remove deprecated module.
ASTERISK-29593
Change-Id: Ib53a42ad974c63871344b95078c61c188e43da99
---
M build_tools/menuselect-deps.in
D channels/chan_oss.c
D configs/samples/oss.conf.sample
M configure
M configure.ac
A doc/UPGRADE-staging/chan_oss_removal.txt
M include/asterisk/autoconfig.h.in
M makeopts.in
M menuselect/configure
9 files changed, 41 insertions(+), 1,952 deletions(-)
Approvals:
Sean Bright: Looks good to me, but someone else must approve
George Joseph: Looks good to me, approved
Joshua Colp: Approved for Submit
diff --git a/build_tools/menuselect-deps.in b/build_tools/menuselect-deps.in
index 161c67b..586985f 100644
--- a/build_tools/menuselect-deps.in
+++ b/build_tools/menuselect-deps.in
@@ -46,7 +46,6 @@
OPUS=@PBX_OPUS@
OPUSFILE=@PBX_OPUSFILE@
OSPTK=@PBX_OSPTK@
-OSS=@PBX_OSS@
PGSQL=@PBX_PGSQL@
PJPROJECT=@PBX_PJPROJECT@
POPT=@PBX_POPT@
diff --git a/channels/chan_oss.c b/channels/chan_oss.c
deleted file mode 100644
index 69dd71f..0000000
--- a/channels/chan_oss.c
+++ /dev/null
@@ -1,1529 +0,0 @@
-/*
- * Asterisk -- An open source telephony toolkit.
- *
- * Copyright (C) 1999 - 2007, Digium, Inc.
- *
- * Mark Spencer <markster at digium.com>
- *
- * FreeBSD changes and multiple device support by Luigi Rizzo, 2005.05.25
- * note-this code best seen with ts=8 (8-spaces tabs) in the editor
- *
- * 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.
- */
-
-// #define HAVE_VIDEO_CONSOLE // uncomment to enable video
-/*! \file
- *
- * \brief Channel driver for OSS sound cards
- *
- * \author Mark Spencer <markster at digium.com>
- * \author Luigi Rizzo
- *
- * \ingroup channel_drivers
- */
-
-/*! \li \ref chan_oss.c uses the configuration file \ref oss.conf
- * \addtogroup configuration_file
- */
-
-/*! \page oss.conf oss.conf
- * \verbinclude oss.conf.sample
- */
-
-/*** MODULEINFO
- <depend>oss</depend>
- <support_level>deprecated</support_level>
- <deprecated_in>16</deprecated_in>
- <removed_in>19</removed_in>
- ***/
-
-#include "asterisk.h"
-
-#include <ctype.h> /* isalnum() used here */
-#include <math.h>
-#include <sys/ioctl.h>
-
-#ifdef __linux
-#include <linux/soundcard.h>
-#elif defined(__FreeBSD__) || defined(__DragonFly__) || defined(__CYGWIN__) || defined(__GLIBC__) || defined(__sun)
-#include <sys/soundcard.h>
-#else
-#include <soundcard.h>
-#endif
-
-#include "asterisk/channel.h"
-#include "asterisk/file.h"
-#include "asterisk/callerid.h"
-#include "asterisk/module.h"
-#include "asterisk/pbx.h"
-#include "asterisk/cli.h"
-#include "asterisk/causes.h"
-#include "asterisk/musiconhold.h"
-#include "asterisk/app.h"
-#include "asterisk/bridge.h"
-#include "asterisk/format_cache.h"
-
-#include "console_video.h"
-
-/*! Global jitterbuffer configuration - by default, jb is disabled
- * \note Values shown here match the defaults shown in oss.conf.sample */
-static struct ast_jb_conf default_jbconf =
-{
- .flags = 0,
- .max_size = 200,
- .resync_threshold = 1000,
- .impl = "fixed",
- .target_extra = 40,
-};
-static struct ast_jb_conf global_jbconf;
-
-/*
- * Basic mode of operation:
- *
- * we have one keyboard (which receives commands from the keyboard)
- * and multiple headset's connected to audio cards.
- * Cards/Headsets are named as the sections of oss.conf.
- * The section called [general] contains the default parameters.
- *
- * At any time, the keyboard is attached to one card, and you
- * can switch among them using the command 'console foo'
- * where 'foo' is the name of the card you want.
- *
- * oss.conf parameters are
-START_CONFIG
-
-[general]
- ; General config options, with default values shown.
- ; You should use one section per device, with [general] being used
- ; for the first device and also as a template for other devices.
- ;
- ; All but 'debug' can go also in the device-specific sections.
- ;
- ; debug = 0x0 ; misc debug flags, default is 0
-
- ; Set the device to use for I/O
- ; device = /dev/dsp
-
- ; Optional mixer command to run upon startup (e.g. to set
- ; volume levels, mutes, etc.
- ; mixer =
-
- ; Software mic volume booster (or attenuator), useful for sound
- ; cards or microphones with poor sensitivity. The volume level
- ; is in dB, ranging from -20.0 to +20.0
- ; boost = n ; mic volume boost in dB
-
- ; Set the callerid for outgoing calls
- ; callerid = John Doe <555-1234>
-
- ; autoanswer = no ; no autoanswer on call
- ; autohangup = yes ; hangup when other party closes
- ; extension = s ; default extension to call
- ; context = default ; default context for outgoing calls
- ; language = "" ; default language
-
- ; Default Music on Hold class to use when this channel is placed on hold in
- ; the case that the music class is not set on the channel with
- ; Set(CHANNEL(musicclass)=whatever) in the dialplan and the peer channel
- ; putting this one on hold did not suggest a class to use.
- ;
- ; mohinterpret=default
-
- ; If you set overridecontext to 'yes', then the whole dial string
- ; will be interpreted as an extension, which is extremely useful
- ; to dial SIP, IAX and other extensions which use the '@' character.
- ; The default is 'no' just for backward compatibility, but the
- ; suggestion is to change it.
- ; overridecontext = no ; if 'no', the last @ will start the context
- ; if 'yes' the whole string is an extension.
-
- ; low level device parameters in case you have problems with the
- ; device driver on your operating system. You should not touch these
- ; unless you know what you are doing.
- ; queuesize = 10 ; frames in device driver
- ; frags = 8 ; argument to SETFRAGMENT
-
- ;------------------------------ JITTER BUFFER CONFIGURATION --------------------------
- ; jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of an
- ; OSS channel. Defaults to "no". An enabled jitterbuffer will
- ; be used only if the sending side can create and the receiving
- ; side can not accept jitter. The OSS channel can't accept jitter,
- ; thus an enabled jitterbuffer on the receive OSS side will always
- ; be used if the sending side can create jitter.
-
- ; jbmaxsize = 200 ; Max length of the jitterbuffer in milliseconds.
-
- ; jbresyncthreshold = 1000 ; Jump in the frame timestamps over which the jitterbuffer is
- ; resynchronized. Useful to improve the quality of the voice, with
- ; big jumps in/broken timestamps, usualy sent from exotic devices
- ; and programs. Defaults to 1000.
-
- ; jbimpl = fixed ; Jitterbuffer implementation, used on the receiving side of an OSS
- ; channel. Two implementations are currenlty available - "fixed"
- ; (with size always equals to jbmax-size) and "adaptive" (with
- ; variable size, actually the new jb of IAX2). Defaults to fixed.
-
- ; jblog = no ; Enables jitterbuffer frame logging. Defaults to "no".
- ;-----------------------------------------------------------------------------------
-
-[card1]
- ; device = /dev/dsp1 ; alternate device
-
-END_CONFIG
-
-.. and so on for the other cards.
-
- */
-
-/*
- * The following parameters are used in the driver:
- *
- * FRAME_SIZE the size of an audio frame, in samples.
- * 160 is used almost universally, so you should not change it.
- *
- * FRAGS the argument for the SETFRAGMENT ioctl.
- * Overridden by the 'frags' parameter in oss.conf
- *
- * Bits 0-7 are the base-2 log of the device's block size,
- * bits 16-31 are the number of blocks in the driver's queue.
- * There are a lot of differences in the way this parameter
- * is supported by different drivers, so you may need to
- * experiment a bit with the value.
- * A good default for linux is 30 blocks of 64 bytes, which
- * results in 6 frames of 320 bytes (160 samples).
- * FreeBSD works decently with blocks of 256 or 512 bytes,
- * leaving the number unspecified.
- * Note that this only refers to the device buffer size,
- * this module will then try to keep the lenght of audio
- * buffered within small constraints.
- *
- * QUEUE_SIZE The max number of blocks actually allowed in the device
- * driver's buffer, irrespective of the available number.
- * Overridden by the 'queuesize' parameter in oss.conf
- *
- * Should be >=2, and at most as large as the hw queue above
- * (otherwise it will never be full).
- */
-
-#define FRAME_SIZE 160
-#define QUEUE_SIZE 10
-
-#if defined(__FreeBSD__)
-#define FRAGS 0x8
-#else
-#define FRAGS ( ( (6 * 5) << 16 ) | 0x6 )
-#endif
-
-/*
- * XXX text message sizes are probably 256 chars, but i am
- * not sure if there is a suitable definition anywhere.
- */
-#define TEXT_SIZE 256
-
-#if 0
-#define TRYOPEN 1 /* try to open on startup */
-#endif
-#define O_CLOSE 0x444 /* special 'close' mode for device */
-/* Which device to use */
-#if defined( __OpenBSD__ ) || defined( __NetBSD__ )
-#define DEV_DSP "/dev/audio"
-#else
-#define DEV_DSP "/dev/dsp"
-#endif
-
-static char *config = "oss.conf"; /* default config file */
-
-static int oss_debug;
-
-/*!
- * \brief descriptor for one of our channels.
- *
- * There is one used for 'default' values (from the [general] entry in
- * the configuration file), and then one instance for each device
- * (the default is cloned from [general], others are only created
- * if the relevant section exists).
- */
-struct chan_oss_pvt {
- struct chan_oss_pvt *next;
-
- char *name;
- int total_blocks; /*!< total blocks in the output device */
- int sounddev;
- enum {
- CHAN_OSS_DUPLEX_UNSET,
- CHAN_OSS_DUPLEX_FULL,
- CHAN_OSS_DUPLEX_READ,
- CHAN_OSS_DUPLEX_WRITE
- } duplex;
- int autoanswer; /*!< Boolean: whether to answer the immediately upon calling */
- int autohangup; /*!< Boolean: whether to hangup the call when the remote end hangs up */
- int hookstate; /*!< Boolean: 1 if offhook; 0 if onhook */
- char *mixer_cmd; /*!< initial command to issue to the mixer */
- unsigned int queuesize; /*!< max fragments in queue */
- unsigned int frags; /*!< parameter for SETFRAGMENT */
-
- int warned; /*!< various flags used for warnings */
-#define WARN_used_blocks 1
-#define WARN_speed 2
-#define WARN_frag 4
- int w_errors; /*!< overfull in the write path */
- struct timeval lastopen;
-
- int overridecontext;
- int mute;
-
- /*! boost support. BOOST_SCALE * 10 ^(BOOST_MAX/20) must
- * be representable in 16 bits to avoid overflows.
- */
-#define BOOST_SCALE (1<<9)
-#define BOOST_MAX 40 /*!< slightly less than 7 bits */
- int boost; /*!< input boost, scaled by BOOST_SCALE */
- char device[64]; /*!< device to open */
-
- pthread_t sthread;
-
- struct ast_channel *owner;
-
- struct video_desc *env; /*!< parameters for video support */
-
- char ext[AST_MAX_EXTENSION];
- char ctx[AST_MAX_CONTEXT];
- char language[MAX_LANGUAGE];
- char cid_name[256]; /*!< Initial CallerID name */
- char cid_num[256]; /*!< Initial CallerID number */
- char mohinterpret[MAX_MUSICCLASS];
-
- /*! buffers used in oss_write */
- char oss_write_buf[FRAME_SIZE * 2];
- int oss_write_dst;
- /*! buffers used in oss_read - AST_FRIENDLY_OFFSET space for headers
- * plus enough room for a full frame
- */
- char oss_read_buf[FRAME_SIZE * 2 + AST_FRIENDLY_OFFSET];
- int readpos; /*!< read position above */
- struct ast_frame read_f; /*!< returned by oss_read */
-};
-
-/*! forward declaration */
-static struct chan_oss_pvt *find_desc(const char *dev);
-
-static char *oss_active; /*!< the active device */
-
-/*! \brief return the pointer to the video descriptor */
-struct video_desc *get_video_desc(struct ast_channel *c)
-{
- struct chan_oss_pvt *o = c ? ast_channel_tech_pvt(c) : find_desc(oss_active);
- return o ? o->env : NULL;
-}
-static struct chan_oss_pvt oss_default = {
- .sounddev = -1,
- .duplex = CHAN_OSS_DUPLEX_UNSET, /* XXX check this */
- .autoanswer = 1,
- .autohangup = 1,
- .queuesize = QUEUE_SIZE,
- .frags = FRAGS,
- .ext = "s",
- .ctx = "default",
- .readpos = AST_FRIENDLY_OFFSET, /* start here on reads */
- .lastopen = { 0, 0 },
- .boost = BOOST_SCALE,
-};
-
-
-static int setformat(struct chan_oss_pvt *o, int mode);
-
-static struct ast_channel *oss_request(const char *type, struct ast_format_cap *cap, const struct ast_assigned_ids *assignedids, const struct ast_channel *requestor,
- const char *data, int *cause);
-static int oss_digit_begin(struct ast_channel *c, char digit);
-static int oss_digit_end(struct ast_channel *c, char digit, unsigned int duration);
-static int oss_text(struct ast_channel *c, const char *text);
-static int oss_hangup(struct ast_channel *c);
-static int oss_answer(struct ast_channel *c);
-static struct ast_frame *oss_read(struct ast_channel *chan);
-static int oss_call(struct ast_channel *c, const char *dest, int timeout);
-static int oss_write(struct ast_channel *chan, struct ast_frame *f);
-static int oss_indicate(struct ast_channel *chan, int cond, const void *data, size_t datalen);
-static int oss_fixup(struct ast_channel *oldchan, struct ast_channel *newchan);
-static char tdesc[] = "OSS Console Channel Driver";
-
-/* cannot do const because need to update some fields at runtime */
-static struct ast_channel_tech oss_tech = {
- .type = "Console",
- .description = tdesc,
- .requester = oss_request,
- .send_digit_begin = oss_digit_begin,
- .send_digit_end = oss_digit_end,
- .send_text = oss_text,
- .hangup = oss_hangup,
- .answer = oss_answer,
- .read = oss_read,
- .call = oss_call,
- .write = oss_write,
- .write_video = console_write_video,
- .indicate = oss_indicate,
- .fixup = oss_fixup,
-};
-
-/*!
- * \brief returns a pointer to the descriptor with the given name
- */
-static struct chan_oss_pvt *find_desc(const char *dev)
-{
- struct chan_oss_pvt *o = NULL;
-
- if (!dev)
- ast_log(LOG_WARNING, "null dev\n");
-
- for (o = oss_default.next; o && o->name && dev && strcmp(o->name, dev) != 0; o = o->next);
-
- if (!o)
- ast_log(LOG_WARNING, "could not find <%s>\n", dev ? dev : "--no-device--");
-
- return o;
-}
-
-/* !
- * \brief split a string in extension-context, returns pointers to malloc'ed
- * strings.
- *
- * If we do not have 'overridecontext' then the last @ is considered as
- * a context separator, and the context is overridden.
- * This is usually not very necessary as you can play with the dialplan,
- * and it is nice not to need it because you have '@' in SIP addresses.
- *
- * \return the buffer address.
- */
-static char *ast_ext_ctx(const char *src, char **ext, char **ctx)
-{
- struct chan_oss_pvt *o = find_desc(oss_active);
-
- if (ext == NULL || ctx == NULL)
- return NULL; /* error */
-
- *ext = *ctx = NULL;
-
- if (src && *src != '\0')
- *ext = ast_strdup(src);
-
- if (*ext == NULL)
- return NULL;
-
- if (!o->overridecontext) {
- /* parse from the right */
- *ctx = strrchr(*ext, '@');
- if (*ctx)
- *(*ctx)++ = '\0';
- }
-
- return *ext;
-}
-
-/*!
- * \brief Returns the number of blocks used in the audio output channel
- */
-static int used_blocks(struct chan_oss_pvt *o)
-{
- struct audio_buf_info info;
-
- if (ioctl(o->sounddev, SNDCTL_DSP_GETOSPACE, &info)) {
- if (!(o->warned & WARN_used_blocks)) {
- ast_log(LOG_WARNING, "Error reading output space\n");
- o->warned |= WARN_used_blocks;
- }
- return 1;
- }
-
- if (o->total_blocks == 0) {
- if (0) /* debugging */
- ast_log(LOG_WARNING, "fragtotal %d size %d avail %d\n", info.fragstotal, info.fragsize, info.fragments);
- o->total_blocks = info.fragments;
- }
-
- return o->total_blocks - info.fragments;
-}
-
-/*! Write an exactly FRAME_SIZE sized frame */
-static int soundcard_writeframe(struct chan_oss_pvt *o, short *data)
-{
- int res;
-
- if (o->sounddev < 0)
- setformat(o, O_RDWR);
- if (o->sounddev < 0)
- return 0; /* not fatal */
- /*
- * Nothing complex to manage the audio device queue.
- * If the buffer is full just drop the extra, otherwise write.
- * XXX in some cases it might be useful to write anyways after
- * a number of failures, to restart the output chain.
- */
- res = used_blocks(o);
- if (res > o->queuesize) { /* no room to write a block */
- if (o->w_errors++ == 0 && (oss_debug & 0x4))
- ast_log(LOG_WARNING, "write: used %d blocks (%d)\n", res, o->w_errors);
- return 0;
- }
- o->w_errors = 0;
- return write(o->sounddev, (void *)data, FRAME_SIZE * 2);
-}
-
-/*!
- * reset and close the device if opened,
- * then open and initialize it in the desired mode,
- * trigger reads and writes so we can start using it.
- */
-static int setformat(struct chan_oss_pvt *o, int mode)
-{
- int fmt, desired, res, fd;
-
- if (o->sounddev >= 0) {
- ioctl(o->sounddev, SNDCTL_DSP_RESET, 0);
- close(o->sounddev);
- o->duplex = CHAN_OSS_DUPLEX_UNSET;
- o->sounddev = -1;
- }
- if (mode == O_CLOSE) /* we are done */
- return 0;
- if (ast_tvdiff_ms(ast_tvnow(), o->lastopen) < 1000)
- return -1; /* don't open too often */
- o->lastopen = ast_tvnow();
- fd = o->sounddev = open(o->device, mode | O_NONBLOCK);
- if (fd < 0) {
- ast_log(LOG_WARNING, "Unable to re-open DSP device %s: %s\n", o->device, strerror(errno));
- return -1;
- }
- if (o->owner)
- ast_channel_set_fd(o->owner, 0, fd);
-
-#if __BYTE_ORDER == __LITTLE_ENDIAN
- fmt = AFMT_S16_LE;
-#else
- fmt = AFMT_S16_BE;
-#endif
- res = ioctl(fd, SNDCTL_DSP_SETFMT, &fmt);
- if (res < 0) {
- ast_log(LOG_WARNING, "Unable to set format to 16-bit signed\n");
- return -1;
- }
- switch (mode) {
- case O_RDWR:
- res = ioctl(fd, SNDCTL_DSP_SETDUPLEX, 0);
- /* Check to see if duplex set (FreeBSD Bug) */
- res = ioctl(fd, SNDCTL_DSP_GETCAPS, &fmt);
- if (res == 0 && (fmt & DSP_CAP_DUPLEX)) {
- ast_verb(2, "Console is full duplex\n");
- o->duplex = CHAN_OSS_DUPLEX_FULL;
- };
- break;
-
- case O_WRONLY:
- o->duplex = CHAN_OSS_DUPLEX_WRITE;
- break;
-
- case O_RDONLY:
- o->duplex = CHAN_OSS_DUPLEX_READ;
- break;
- }
-
- fmt = 0;
- res = ioctl(fd, SNDCTL_DSP_STEREO, &fmt);
- if (res < 0) {
- ast_log(LOG_WARNING, "Failed to set audio device to mono\n");
- return -1;
- }
- fmt = desired = DEFAULT_SAMPLE_RATE; /* 8000 Hz desired */
- res = ioctl(fd, SNDCTL_DSP_SPEED, &fmt);
-
- if (res < 0) {
- ast_log(LOG_WARNING, "Failed to set sample rate to %d\n", desired);
- return -1;
- }
- if (fmt != desired) {
- if (!(o->warned & WARN_speed)) {
- ast_log(LOG_WARNING,
- "Requested %d Hz, got %d Hz -- sound may be choppy\n",
- desired, fmt);
- o->warned |= WARN_speed;
- }
- }
- /*
- * on Freebsd, SETFRAGMENT does not work very well on some cards.
- * Default to use 256 bytes, let the user override
- */
- if (o->frags) {
- fmt = o->frags;
- res = ioctl(fd, SNDCTL_DSP_SETFRAGMENT, &fmt);
- if (res < 0) {
- if (!(o->warned & WARN_frag)) {
- ast_log(LOG_WARNING,
- "Unable to set fragment size -- sound may be choppy\n");
- o->warned |= WARN_frag;
- }
- }
- }
- /* on some cards, we need SNDCTL_DSP_SETTRIGGER to start outputting */
- res = PCM_ENABLE_INPUT | PCM_ENABLE_OUTPUT;
- res = ioctl(fd, SNDCTL_DSP_SETTRIGGER, &res);
- /* it may fail if we are in half duplex, never mind */
- return 0;
-}
-
-/*
- * some of the standard methods supported by channels.
- */
-static int oss_digit_begin(struct ast_channel *c, char digit)
-{
- return 0;
-}
-
-static int oss_digit_end(struct ast_channel *c, char digit, unsigned int duration)
-{
- /* no better use for received digits than print them */
- ast_verbose(" << Console Received digit %c of duration %u ms >> \n",
- digit, duration);
- return 0;
-}
-
-static int oss_text(struct ast_channel *c, const char *text)
-{
- /* print received messages */
- ast_verbose(" << Console Received text %s >> \n", text);
- return 0;
-}
-
-/*!
- * \brief handler for incoming calls. Either autoanswer, or start ringing
- */
-static int oss_call(struct ast_channel *c, const char *dest, int timeout)
-{
- struct chan_oss_pvt *o = ast_channel_tech_pvt(c);
- struct ast_frame f = { AST_FRAME_CONTROL, };
- AST_DECLARE_APP_ARGS(args,
- AST_APP_ARG(name);
- AST_APP_ARG(flags);
- );
- char *parse = ast_strdupa(dest);
-
- AST_NONSTANDARD_APP_ARGS(args, parse, '/');
-
- ast_verbose(" << Call to device '%s' dnid '%s' rdnis '%s' on console from '%s' <%s> >>\n",
- dest,
- S_OR(ast_channel_dialed(c)->number.str, ""),
- S_COR(ast_channel_redirecting(c)->from.number.valid, ast_channel_redirecting(c)->from.number.str, ""),
- S_COR(ast_channel_caller(c)->id.name.valid, ast_channel_caller(c)->id.name.str, ""),
- S_COR(ast_channel_caller(c)->id.number.valid, ast_channel_caller(c)->id.number.str, ""));
- if (!ast_strlen_zero(args.flags) && strcasecmp(args.flags, "answer") == 0) {
- f.subclass.integer = AST_CONTROL_ANSWER;
- ast_queue_frame(c, &f);
- } else if (!ast_strlen_zero(args.flags) && strcasecmp(args.flags, "noanswer") == 0) {
- f.subclass.integer = AST_CONTROL_RINGING;
- ast_queue_frame(c, &f);
- ast_indicate(c, AST_CONTROL_RINGING);
- } else if (o->autoanswer) {
- ast_verbose(" << Auto-answered >> \n");
- f.subclass.integer = AST_CONTROL_ANSWER;
- ast_queue_frame(c, &f);
- o->hookstate = 1;
- } else {
- ast_verbose("<< Type 'answer' to answer, or use 'autoanswer' for future calls >> \n");
- f.subclass.integer = AST_CONTROL_RINGING;
- ast_queue_frame(c, &f);
- ast_indicate(c, AST_CONTROL_RINGING);
- }
- return 0;
-}
-
-/*!
- * \brief remote side answered the phone
- */
-static int oss_answer(struct ast_channel *c)
-{
- struct chan_oss_pvt *o = ast_channel_tech_pvt(c);
- ast_verbose(" << Console call has been answered >> \n");
- ast_setstate(c, AST_STATE_UP);
- o->hookstate = 1;
- return 0;
-}
-
-static int oss_hangup(struct ast_channel *c)
-{
- struct chan_oss_pvt *o = ast_channel_tech_pvt(c);
-
- ast_channel_tech_pvt_set(c, NULL);
- o->owner = NULL;
- ast_verbose(" << Hangup on console >> \n");
- console_video_uninit(o->env);
- ast_module_unref(ast_module_info->self);
- if (o->hookstate) {
- if (o->autoanswer || o->autohangup) {
- /* Assume auto-hangup too */
- o->hookstate = 0;
- setformat(o, O_CLOSE);
- }
- }
- return 0;
-}
-
-/*! \brief used for data coming from the network */
-static int oss_write(struct ast_channel *c, struct ast_frame *f)
-{
- int src;
- struct chan_oss_pvt *o = ast_channel_tech_pvt(c);
-
- /*
- * we could receive a block which is not a multiple of our
- * FRAME_SIZE, so buffer it locally and write to the device
- * in FRAME_SIZE chunks.
- * Keep the residue stored for future use.
- */
- src = 0; /* read position into f->data */
- while (src < f->datalen) {
- /* Compute spare room in the buffer */
- int l = sizeof(o->oss_write_buf) - o->oss_write_dst;
-
- if (f->datalen - src >= l) { /* enough to fill a frame */
- memcpy(o->oss_write_buf + o->oss_write_dst, f->data.ptr + src, l);
- soundcard_writeframe(o, (short *) o->oss_write_buf);
- src += l;
- o->oss_write_dst = 0;
- } else { /* copy residue */
- l = f->datalen - src;
- memcpy(o->oss_write_buf + o->oss_write_dst, f->data.ptr + src, l);
- src += l; /* but really, we are done */
- o->oss_write_dst += l;
- }
- }
- return 0;
-}
-
-static struct ast_frame *oss_read(struct ast_channel *c)
-{
- int res;
- struct chan_oss_pvt *o = ast_channel_tech_pvt(c);
- struct ast_frame *f = &o->read_f;
-
- /* XXX can be simplified returning &ast_null_frame */
- /* prepare a NULL frame in case we don't have enough data to return */
- memset(f, '\0', sizeof(struct ast_frame));
- f->frametype = AST_FRAME_NULL;
- f->src = oss_tech.type;
-
- res = read(o->sounddev, o->oss_read_buf + o->readpos, sizeof(o->oss_read_buf) - o->readpos);
- if (res < 0) /* audio data not ready, return a NULL frame */
- return f;
-
- o->readpos += res;
- if (o->readpos < sizeof(o->oss_read_buf)) /* not enough samples */
- return f;
-
- if (o->mute)
- return f;
-
- o->readpos = AST_FRIENDLY_OFFSET; /* reset read pointer for next frame */
- if (ast_channel_state(c) != AST_STATE_UP) /* drop data if frame is not up */
- return f;
- /* ok we can build and deliver the frame to the caller */
- f->frametype = AST_FRAME_VOICE;
- f->subclass.format = ast_format_slin;
- f->samples = FRAME_SIZE;
- f->datalen = FRAME_SIZE * 2;
- f->data.ptr = o->oss_read_buf + AST_FRIENDLY_OFFSET;
- if (o->boost != BOOST_SCALE) { /* scale and clip values */
- int i, x;
- int16_t *p = (int16_t *) f->data.ptr;
- for (i = 0; i < f->samples; i++) {
- x = (p[i] * o->boost) / BOOST_SCALE;
- if (x > 32767)
- x = 32767;
- else if (x < -32768)
- x = -32768;
- p[i] = x;
- }
- }
-
- f->offset = AST_FRIENDLY_OFFSET;
- return f;
-}
-
-static int oss_fixup(struct ast_channel *oldchan, struct ast_channel *newchan)
-{
- struct chan_oss_pvt *o = ast_channel_tech_pvt(newchan);
- o->owner = newchan;
- return 0;
-}
-
-static int oss_indicate(struct ast_channel *c, int cond, const void *data, size_t datalen)
-{
- struct chan_oss_pvt *o = ast_channel_tech_pvt(c);
- int res = 0;
-
- switch (cond) {
- case AST_CONTROL_INCOMPLETE:
- case AST_CONTROL_BUSY:
- case AST_CONTROL_CONGESTION:
- case AST_CONTROL_RINGING:
- case AST_CONTROL_PVT_CAUSE_CODE:
- case -1:
- res = -1;
- break;
- case AST_CONTROL_PROGRESS:
- case AST_CONTROL_PROCEEDING:
- case AST_CONTROL_VIDUPDATE:
- case AST_CONTROL_SRCUPDATE:
- break;
- case AST_CONTROL_HOLD:
- ast_verbose(" << Console Has Been Placed on Hold >> \n");
- ast_moh_start(c, data, o->mohinterpret);
- break;
- case AST_CONTROL_UNHOLD:
- ast_verbose(" << Console Has Been Retrieved from Hold >> \n");
- ast_moh_stop(c);
- break;
- default:
- ast_log(LOG_WARNING, "Don't know how to display condition %d on %s\n", cond, ast_channel_name(c));
- return -1;
- }
-
- return res;
-}
-
-/*!
- * \brief allocate a new channel.
- */
-static struct ast_channel *oss_new(struct chan_oss_pvt *o, char *ext, char *ctx, int state, const struct ast_assigned_ids *assignedids, const struct ast_channel *requestor)
-{
- struct ast_channel *c;
-
- c = ast_channel_alloc(1, state, o->cid_num, o->cid_name, "", ext, ctx, assignedids, requestor, 0, "Console/%s", o->device + 5);
- if (c == NULL)
- return NULL;
- ast_channel_tech_set(c, &oss_tech);
- if (o->sounddev < 0)
- setformat(o, O_RDWR);
- ast_channel_set_fd(c, 0, o->sounddev); /* -1 if device closed, override later */
-
- ast_channel_set_readformat(c, ast_format_slin);
- ast_channel_set_writeformat(c, ast_format_slin);
- ast_channel_nativeformats_set(c, oss_tech.capabilities);
-
- /* if the console makes the call, add video to the offer */
- /* if (state == AST_STATE_RINGING) TODO XXX CONSOLE VIDEO IS DISABLED UNTIL IT GETS A MAINTAINER
- c->nativeformats |= console_video_formats; */
-
- ast_channel_tech_pvt_set(c, o);
-
- if (!ast_strlen_zero(o->language))
- ast_channel_language_set(c, o->language);
- /* Don't use ast_set_callerid() here because it will
- * generate a needless NewCallerID event */
- if (!ast_strlen_zero(o->cid_num)) {
- ast_channel_caller(c)->ani.number.valid = 1;
- ast_channel_caller(c)->ani.number.str = ast_strdup(o->cid_num);
- }
- if (!ast_strlen_zero(ext)) {
- ast_channel_dialed(c)->number.str = ast_strdup(ext);
- }
-
- o->owner = c;
- ast_module_ref(ast_module_info->self);
- ast_jb_configure(c, &global_jbconf);
- ast_channel_unlock(c);
- if (state != AST_STATE_DOWN) {
- if (ast_pbx_start(c)) {
- ast_log(LOG_WARNING, "Unable to start PBX on %s\n", ast_channel_name(c));
- ast_hangup(c);
- o->owner = c = NULL;
- }
- }
- console_video_start(get_video_desc(c), c); /* XXX cleanup */
-
- return c;
-}
-
-static struct ast_channel *oss_request(const char *type, struct ast_format_cap *cap, const struct ast_assigned_ids *assignedids, const struct ast_channel *requestor, const char *data, int *cause)
-{
- struct ast_channel *c;
- struct chan_oss_pvt *o;
- AST_DECLARE_APP_ARGS(args,
- AST_APP_ARG(name);
- AST_APP_ARG(flags);
- );
- char *parse = ast_strdupa(data);
-
- AST_NONSTANDARD_APP_ARGS(args, parse, '/');
- o = find_desc(args.name);
-
- ast_log(LOG_WARNING, "oss_request ty <%s> data 0x%p <%s>\n", type, data, data);
- if (o == NULL) {
- ast_log(LOG_NOTICE, "Device %s not found\n", args.name);
- /* XXX we could default to 'dsp' perhaps ? */
- return NULL;
- }
- if (ast_format_cap_iscompatible_format(cap, ast_format_slin) == AST_FORMAT_CMP_NOT_EQUAL) {
- struct ast_str *codec_buf = ast_str_alloca(AST_FORMAT_CAP_NAMES_LEN);
- ast_log(LOG_NOTICE, "Format %s unsupported\n", ast_format_cap_get_names(cap, &codec_buf));
- return NULL;
- }
- if (o->owner) {
- ast_log(LOG_NOTICE, "Already have a call (chan %p) on the OSS channel\n", o->owner);
- *cause = AST_CAUSE_BUSY;
- return NULL;
- }
- c = oss_new(o, NULL, NULL, AST_STATE_DOWN, assignedids, requestor);
- if (c == NULL) {
- ast_log(LOG_WARNING, "Unable to create new OSS channel\n");
- return NULL;
- }
- return c;
-}
-
-static void store_config_core(struct chan_oss_pvt *o, const char *var, const char *value);
-
-/*! Generic console command handler. Basically a wrapper for a subset
- * of config file options which are also available from the CLI
- */
-static char *console_cmd(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
-{
- struct chan_oss_pvt *o = find_desc(oss_active);
- const char *var, *value;
- switch (cmd) {
- case CLI_INIT:
- e->command = CONSOLE_VIDEO_CMDS;
- e->usage =
- "Usage: " CONSOLE_VIDEO_CMDS "...\n"
- " Generic handler for console commands.\n";
- return NULL;
-
- case CLI_GENERATE:
- return NULL;
- }
-
- if (a->argc < e->args)
- return CLI_SHOWUSAGE;
- if (o == NULL) {
- ast_log(LOG_WARNING, "Cannot find device %s (should not happen!)\n",
- oss_active);
- return CLI_FAILURE;
- }
- var = a->argv[e->args-1];
- value = a->argc > e->args ? a->argv[e->args] : NULL;
- if (value) /* handle setting */
- store_config_core(o, var, value);
- if (!console_video_cli(o->env, var, a->fd)) /* print video-related values */
- return CLI_SUCCESS;
- /* handle other values */
- if (!strcasecmp(var, "device")) {
- ast_cli(a->fd, "device is [%s]\n", o->device);
- }
- return CLI_SUCCESS;
-}
-
-static char *console_autoanswer(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
-{
- struct chan_oss_pvt *o = find_desc(oss_active);
-
- switch (cmd) {
- case CLI_INIT:
- e->command = "console {set|show} autoanswer [on|off]";
- e->usage =
- "Usage: console {set|show} autoanswer [on|off]\n"
- " Enables or disables autoanswer feature. If used without\n"
- " argument, displays the current on/off status of autoanswer.\n"
- " The default value of autoanswer is in 'oss.conf'.\n";
- return NULL;
-
- case CLI_GENERATE:
- return NULL;
- }
-
- if (a->argc == e->args - 1) {
- ast_cli(a->fd, "Auto answer is %s.\n", o->autoanswer ? "on" : "off");
- return CLI_SUCCESS;
- }
- if (a->argc != e->args)
- return CLI_SHOWUSAGE;
- if (o == NULL) {
- ast_log(LOG_WARNING, "Cannot find device %s (should not happen!)\n",
- oss_active);
- return CLI_FAILURE;
- }
- if (!strcasecmp(a->argv[e->args-1], "on"))
- o->autoanswer = 1;
- else if (!strcasecmp(a->argv[e->args - 1], "off"))
- o->autoanswer = 0;
- else
- return CLI_SHOWUSAGE;
- return CLI_SUCCESS;
-}
-
-/*! \brief helper function for the answer key/cli command */
-static char *console_do_answer(int fd)
-{
- struct ast_frame f = { AST_FRAME_CONTROL, { AST_CONTROL_ANSWER } };
- struct chan_oss_pvt *o = find_desc(oss_active);
- if (!o->owner) {
- if (fd > -1)
- ast_cli(fd, "No one is calling us\n");
- return CLI_FAILURE;
- }
- o->hookstate = 1;
- ast_queue_frame(o->owner, &f);
- return CLI_SUCCESS;
-}
-
-/*!
- * \brief answer command from the console
- */
-static char *console_answer(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
-{
- switch (cmd) {
- case CLI_INIT:
- e->command = "console answer";
- e->usage =
- "Usage: console answer\n"
- " Answers an incoming call on the console (OSS) channel.\n";
- return NULL;
-
- case CLI_GENERATE:
- return NULL; /* no completion */
- }
- if (a->argc != e->args)
- return CLI_SHOWUSAGE;
- return console_do_answer(a->fd);
-}
-
-/*!
- * \brief Console send text CLI command
- *
- * \note concatenate all arguments into a single string. argv is NULL-terminated
- * so we can use it right away
- */
-static char *console_sendtext(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
-{
- struct chan_oss_pvt *o = find_desc(oss_active);
- char buf[TEXT_SIZE];
-
- if (cmd == CLI_INIT) {
- e->command = "console send text";
- e->usage =
- "Usage: console send text <message>\n"
- " Sends a text message for display on the remote terminal.\n";
- return NULL;
- } else if (cmd == CLI_GENERATE)
- return NULL;
-
- if (a->argc < e->args + 1)
- return CLI_SHOWUSAGE;
- if (!o->owner) {
- ast_cli(a->fd, "Not in a call\n");
- return CLI_FAILURE;
- }
- ast_join(buf, sizeof(buf) - 1, a->argv + e->args);
- if (!ast_strlen_zero(buf)) {
- struct ast_frame f = { 0, };
- int i = strlen(buf);
- buf[i] = '\n';
- f.frametype = AST_FRAME_TEXT;
- f.subclass.integer = 0;
- f.data.ptr = buf;
- f.datalen = i + 1;
- ast_queue_frame(o->owner, &f);
- }
- return CLI_SUCCESS;
-}
-
-static char *console_hangup(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
-{
- struct chan_oss_pvt *o = find_desc(oss_active);
-
- if (cmd == CLI_INIT) {
- e->command = "console hangup";
- e->usage =
- "Usage: console hangup\n"
- " Hangs up any call currently placed on the console.\n";
- return NULL;
- } else if (cmd == CLI_GENERATE)
- return NULL;
-
- if (a->argc != e->args)
- return CLI_SHOWUSAGE;
- if (!o->owner && !o->hookstate) { /* XXX maybe only one ? */
- ast_cli(a->fd, "No call to hang up\n");
- return CLI_FAILURE;
- }
- o->hookstate = 0;
- if (o->owner)
- ast_queue_hangup_with_cause(o->owner, AST_CAUSE_NORMAL_CLEARING);
- setformat(o, O_CLOSE);
- return CLI_SUCCESS;
-}
-
-static char *console_flash(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
-{
- struct ast_frame f = { AST_FRAME_CONTROL, { AST_CONTROL_FLASH } };
- struct chan_oss_pvt *o = find_desc(oss_active);
-
- if (cmd == CLI_INIT) {
- e->command = "console flash";
- e->usage =
- "Usage: console flash\n"
- " Flashes the call currently placed on the console.\n";
- return NULL;
- } else if (cmd == CLI_GENERATE)
- return NULL;
-
- if (a->argc != e->args)
- return CLI_SHOWUSAGE;
- if (!o->owner) { /* XXX maybe !o->hookstate too ? */
- ast_cli(a->fd, "No call to flash\n");
- return CLI_FAILURE;
- }
- o->hookstate = 0;
- if (o->owner)
- ast_queue_frame(o->owner, &f);
- return CLI_SUCCESS;
-}
-
-static char *console_dial(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
-{
- char *s = NULL;
- char *mye = NULL, *myc = NULL;
- struct chan_oss_pvt *o = find_desc(oss_active);
-
- if (cmd == CLI_INIT) {
- e->command = "console dial";
- e->usage =
- "Usage: console dial [extension[@context]]\n"
- " Dials a given extension (and context if specified)\n";
- return NULL;
- } else if (cmd == CLI_GENERATE)
- return NULL;
-
- if (a->argc > e->args + 1)
- return CLI_SHOWUSAGE;
- if (o->owner) { /* already in a call */
- int i;
- struct ast_frame f = { AST_FRAME_DTMF, { 0 } };
- const char *digits;
-
- if (a->argc == e->args) { /* argument is mandatory here */
- ast_cli(a->fd, "Already in a call. You can only dial digits until you hangup.\n");
- return CLI_FAILURE;
- }
- digits = a->argv[e->args];
- /* send the string one char at a time */
- for (i = 0; i < strlen(digits); i++) {
- f.subclass.integer = digits[i];
- ast_queue_frame(o->owner, &f);
- }
- return CLI_SUCCESS;
- }
- /* if we have an argument split it into extension and context */
- if (a->argc == e->args + 1)
- s = ast_ext_ctx(a->argv[e->args], &mye, &myc);
- /* supply default values if needed */
- if (mye == NULL)
- mye = o->ext;
- if (myc == NULL)
- myc = o->ctx;
- if (ast_exists_extension(NULL, myc, mye, 1, NULL)) {
- o->hookstate = 1;
- oss_new(o, mye, myc, AST_STATE_RINGING, NULL, NULL);
- } else
- ast_cli(a->fd, "No such extension '%s' in context '%s'\n", mye, myc);
- if (s)
- ast_free(s);
- return CLI_SUCCESS;
-}
-
-static char *console_mute(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
-{
- struct chan_oss_pvt *o = find_desc(oss_active);
- const char *s;
- int toggle = 0;
-
- if (cmd == CLI_INIT) {
- e->command = "console {mute|unmute} [toggle]";
- e->usage =
- "Usage: console {mute|unmute} [toggle]\n"
- " Mute/unmute the microphone.\n";
- return NULL;
- } else if (cmd == CLI_GENERATE)
- return NULL;
-
- if (a->argc > e->args)
- return CLI_SHOWUSAGE;
- if (a->argc == e->args) {
- if (strcasecmp(a->argv[e->args-1], "toggle"))
- return CLI_SHOWUSAGE;
- toggle = 1;
- }
- s = a->argv[e->args-2];
- if (!strcasecmp(s, "mute"))
- o->mute = toggle ? !o->mute : 1;
- else if (!strcasecmp(s, "unmute"))
- o->mute = toggle ? !o->mute : 0;
- else
- return CLI_SHOWUSAGE;
- ast_cli(a->fd, "Console mic is %s\n", o->mute ? "off" : "on");
- return CLI_SUCCESS;
-}
-
-static char *console_transfer(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
-{
- struct chan_oss_pvt *o = find_desc(oss_active);
- char *tmp, *ext, *ctx;
-
- switch (cmd) {
- case CLI_INIT:
- e->command = "console transfer";
- e->usage =
- "Usage: console transfer <extension>[@context]\n"
- " Transfers the currently connected call to the given extension (and\n"
- " context if specified)\n";
- return NULL;
- case CLI_GENERATE:
- return NULL;
- }
-
- if (a->argc != 3)
- return CLI_SHOWUSAGE;
- if (o == NULL)
- return CLI_FAILURE;
- if (o->owner == NULL || !ast_channel_is_bridged(o->owner)) {
- ast_cli(a->fd, "There is no call to transfer\n");
- return CLI_SUCCESS;
- }
-
- tmp = ast_ext_ctx(a->argv[2], &ext, &ctx);
- if (ctx == NULL) { /* supply default context if needed */
- ctx = ast_strdupa(ast_channel_context(o->owner));
- }
- if (ast_bridge_transfer_blind(1, o->owner, ext, ctx, NULL, NULL) != AST_BRIDGE_TRANSFER_SUCCESS) {
- ast_log(LOG_WARNING, "Unable to transfer call from channel %s\n", ast_channel_name(o->owner));
- }
- ast_free(tmp);
- return CLI_SUCCESS;
-}
-
-static char *console_active(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
-{
- switch (cmd) {
- case CLI_INIT:
- e->command = "console {set|show} active [<device>]";
- e->usage =
- "Usage: console active [device]\n"
- " If used without a parameter, displays which device is the current\n"
- " console. If a device is specified, the console sound device is changed to\n"
- " the device specified.\n";
- return NULL;
- case CLI_GENERATE:
- return NULL;
- }
-
- if (a->argc == 3)
- ast_cli(a->fd, "active console is [%s]\n", oss_active);
- else if (a->argc != 4)
- return CLI_SHOWUSAGE;
- else {
- struct chan_oss_pvt *o;
- if (strcmp(a->argv[3], "show") == 0) {
- for (o = oss_default.next; o; o = o->next)
- ast_cli(a->fd, "device [%s] exists\n", o->name);
- return CLI_SUCCESS;
- }
- o = find_desc(a->argv[3]);
- if (o == NULL)
- ast_cli(a->fd, "No device [%s] exists\n", a->argv[3]);
- else
- oss_active = o->name;
- }
- return CLI_SUCCESS;
-}
-
-/*!
- * \brief store the boost factor
- */
-static void store_boost(struct chan_oss_pvt *o, const char *s)
-{
- double boost = 0;
- if (sscanf(s, "%30lf", &boost) != 1) {
- ast_log(LOG_WARNING, "invalid boost <%s>\n", s);
- return;
- }
- if (boost < -BOOST_MAX) {
- ast_log(LOG_WARNING, "boost %s too small, using %d\n", s, -BOOST_MAX);
- boost = -BOOST_MAX;
- } else if (boost > BOOST_MAX) {
- ast_log(LOG_WARNING, "boost %s too large, using %d\n", s, BOOST_MAX);
- boost = BOOST_MAX;
- }
- boost = exp(log(10) * boost / 20) * BOOST_SCALE;
- o->boost = boost;
- ast_log(LOG_WARNING, "setting boost %s to %d\n", s, o->boost);
-}
-
-static char *console_boost(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
-{
- struct chan_oss_pvt *o = find_desc(oss_active);
-
- switch (cmd) {
- case CLI_INIT:
- e->command = "console boost";
- e->usage =
- "Usage: console boost [boost in dB]\n"
- " Sets or display mic boost in dB\n";
- return NULL;
- case CLI_GENERATE:
- return NULL;
- }
-
- if (a->argc == 2)
- ast_cli(a->fd, "boost currently %5.1f\n", 20 * log10(((double) o->boost / (double) BOOST_SCALE)));
- else if (a->argc == 3)
- store_boost(o, a->argv[2]);
- return CLI_SUCCESS;
-}
-
-static struct ast_cli_entry cli_oss[] = {
- AST_CLI_DEFINE(console_answer, "Answer an incoming console call"),
- AST_CLI_DEFINE(console_hangup, "Hangup a call on the console"),
- AST_CLI_DEFINE(console_flash, "Flash a call on the console"),
- AST_CLI_DEFINE(console_dial, "Dial an extension on the console"),
- AST_CLI_DEFINE(console_mute, "Disable/Enable mic input"),
- AST_CLI_DEFINE(console_transfer, "Transfer a call to a different extension"),
- AST_CLI_DEFINE(console_cmd, "Generic console command"),
- AST_CLI_DEFINE(console_sendtext, "Send text to the remote device"),
- AST_CLI_DEFINE(console_autoanswer, "Sets/displays autoanswer"),
- AST_CLI_DEFINE(console_boost, "Sets/displays mic boost in dB"),
- AST_CLI_DEFINE(console_active, "Sets/displays active console"),
-};
-
-/*!
- * store the mixer argument from the config file, filtering possibly
- * invalid or dangerous values (the string is used as argument for
- * system("mixer %s")
- */
-static void store_mixer(struct chan_oss_pvt *o, const char *s)
-{
- int i;
-
- for (i = 0; i < strlen(s); i++) {
- if (!isalnum(s[i]) && strchr(" \t-/", s[i]) == NULL) {
- ast_log(LOG_WARNING, "Suspect char %c in mixer cmd, ignoring:\n\t%s\n", s[i], s);
- return;
- }
- }
- if (o->mixer_cmd)
- ast_free(o->mixer_cmd);
- o->mixer_cmd = ast_strdup(s);
- ast_log(LOG_WARNING, "setting mixer %s\n", s);
-}
-
-/*!
- * store the callerid components
- */
-static void store_callerid(struct chan_oss_pvt *o, const char *s)
-{
- ast_callerid_split(s, o->cid_name, sizeof(o->cid_name), o->cid_num, sizeof(o->cid_num));
-}
-
-static void store_config_core(struct chan_oss_pvt *o, const char *var, const char *value)
-{
- CV_START(var, value);
-
- /* handle jb conf */
- if (!ast_jb_read_conf(&global_jbconf, var, value))
- return;
-
- if (!console_video_config(&o->env, var, value))
- return; /* matched there */
- CV_BOOL("autoanswer", o->autoanswer);
- CV_BOOL("autohangup", o->autohangup);
- CV_BOOL("overridecontext", o->overridecontext);
- CV_STR("device", o->device);
- CV_UINT("frags", o->frags);
- CV_UINT("debug", oss_debug);
- CV_UINT("queuesize", o->queuesize);
- CV_STR("context", o->ctx);
- CV_STR("language", o->language);
- CV_STR("mohinterpret", o->mohinterpret);
- CV_STR("extension", o->ext);
- CV_F("mixer", store_mixer(o, value));
- CV_F("callerid", store_callerid(o, value)) ;
- CV_F("boost", store_boost(o, value));
-
- CV_END;
-}
-
-/*!
- * grab fields from the config file, init the descriptor and open the device.
- */
-static struct chan_oss_pvt *store_config(struct ast_config *cfg, char *ctg)
-{
- struct ast_variable *v;
- struct chan_oss_pvt *o;
-
- if (ctg == NULL) {
- o = &oss_default;
- ctg = "general";
- } else {
- if (!(o = ast_calloc(1, sizeof(*o))))
- return NULL;
- *o = oss_default;
- /* "general" is also the default thing */
- if (strcmp(ctg, "general") == 0) {
- o->name = ast_strdup("dsp");
- oss_active = o->name;
- goto openit;
- }
- o->name = ast_strdup(ctg);
- }
-
- strcpy(o->mohinterpret, "default");
-
- o->lastopen = ast_tvnow(); /* don't leave it 0 or tvdiff may wrap */
- /* fill other fields from configuration */
- for (v = ast_variable_browse(cfg, ctg); v; v = v->next) {
- store_config_core(o, v->name, v->value);
- }
- if (ast_strlen_zero(o->device))
- ast_copy_string(o->device, DEV_DSP, sizeof(o->device));
- if (o->mixer_cmd) {
- char *cmd;
-
- if (ast_asprintf(&cmd, "mixer %s", o->mixer_cmd) >= 0) {
- ast_log(LOG_WARNING, "running [%s]\n", cmd);
- if (system(cmd) < 0) {
- ast_log(LOG_WARNING, "system() failed: %s\n", strerror(errno));
- }
- ast_free(cmd);
- }
- }
-
- /* if the config file requested to start the GUI, do it */
- if (get_gui_startup(o->env))
- console_video_start(o->env, NULL);
-
- if (o == &oss_default) /* we are done with the default */
- return NULL;
-
-openit:
-#ifdef TRYOPEN
- if (setformat(o, O_RDWR) < 0) { /* open device */
- ast_verb(1, "Device %s not detected\n", ctg);
- ast_verb(1, "Turn off OSS support by adding " "'noload=chan_oss.so' in /etc/asterisk/modules.conf\n");
- goto error;
- }
- if (o->duplex != CHAN_OSS_DUPLEX_FULL)
- ast_log(LOG_WARNING, "XXX I don't work right with non " "full-duplex sound cards XXX\n");
-#endif /* TRYOPEN */
-
- /* link into list of devices */
- if (o != &oss_default) {
- o->next = oss_default.next;
- oss_default.next = o;
- }
- return o;
-
-#ifdef TRYOPEN
-error:
- if (o != &oss_default)
- ast_free(o);
- return NULL;
-#endif
-}
-
-static int unload_module(void)
-{
- struct chan_oss_pvt *o, *next;
-
- ast_channel_unregister(&oss_tech);
- ast_cli_unregister_multiple(cli_oss, ARRAY_LEN(cli_oss));
-
- o = oss_default.next;
- while (o) {
- close(o->sounddev);
- if (o->owner)
- ast_softhangup(o->owner, AST_SOFTHANGUP_APPUNLOAD);
- if (o->owner)
- return -1;
- next = o->next;
- ast_free(o->name);
- ast_free(o);
- o = next;
- }
- ao2_cleanup(oss_tech.capabilities);
- oss_tech.capabilities = NULL;
-
- return 0;
-}
-
-/*!
- * \brief Load the module
- *
- * Module loading including tests for configuration or dependencies.
- * This function can return AST_MODULE_LOAD_FAILURE, AST_MODULE_LOAD_DECLINE,
- * or AST_MODULE_LOAD_SUCCESS. If a dependency or environment variable fails
- * tests return AST_MODULE_LOAD_FAILURE. If the module can not load the
- * configuration file or other non-critical problem return
- * AST_MODULE_LOAD_DECLINE. On success return AST_MODULE_LOAD_SUCCESS.
- */
-static int load_module(void)
-{
- struct ast_config *cfg = NULL;
- char *ctg = NULL;
- struct ast_flags config_flags = { 0 };
-
- /* Copy the default jb config over global_jbconf */
- memcpy(&global_jbconf, &default_jbconf, sizeof(struct ast_jb_conf));
-
- /* load config file */
- if (!(cfg = ast_config_load(config, config_flags))) {
- ast_log(LOG_NOTICE, "Unable to load config %s\n", config);
- return AST_MODULE_LOAD_DECLINE;
- } else if (cfg == CONFIG_STATUS_FILEINVALID) {
- ast_log(LOG_ERROR, "Config file %s is in an invalid format. Aborting.\n", config);
- return AST_MODULE_LOAD_DECLINE;
- }
-
- do {
- store_config(cfg, ctg);
- } while ( (ctg = ast_category_browse(cfg, ctg)) != NULL);
-
- ast_config_destroy(cfg);
-
- if (find_desc(oss_active) == NULL) {
- ast_log(LOG_NOTICE, "Device %s not found\n", oss_active);
- /* XXX we could default to 'dsp' perhaps ? */
- unload_module();
- return AST_MODULE_LOAD_DECLINE;
- }
-
- if (!(oss_tech.capabilities = ast_format_cap_alloc(0))) {
- return AST_MODULE_LOAD_DECLINE;
- }
- ast_format_cap_append(oss_tech.capabilities, ast_format_slin, 0);
-
- /* TODO XXX CONSOLE VIDEO IS DISABLE UNTIL IT HAS A MAINTAINER
- * add console_video_formats to oss_tech.capabilities once this occurs. */
-
- if (ast_channel_register(&oss_tech)) {
- ast_log(LOG_ERROR, "Unable to register channel type 'OSS'\n");
- return AST_MODULE_LOAD_DECLINE;
- }
-
- ast_cli_register_multiple(cli_oss, ARRAY_LEN(cli_oss));
-
- return AST_MODULE_LOAD_SUCCESS;
-}
-
-AST_MODULE_INFO_STANDARD_DEPRECATED(ASTERISK_GPL_KEY, "OSS Console Channel Driver");
diff --git a/configs/samples/oss.conf.sample b/configs/samples/oss.conf.sample
deleted file mode 100644
index b0b3831..0000000
--- a/configs/samples/oss.conf.sample
+++ /dev/null
@@ -1,152 +0,0 @@
-;
-; Automatically generated from ../channels/chan_oss.c
-;
-
-[general]
- ; General config options, with default values shown.
- ; You should use one section per device, with [general] being used
- ; for the first device and also as a template for other devices.
- ;
- ; All but 'debug' can go also in the device-specific sections.
- ;
- ; debug = 0x0 ; misc debug flags, default is 0
-
- ; Set the device to use for I/O
- ; device = /dev/dsp
-
- ; Optional mixer command to run upon startup (e.g. to set
- ; volume levels, mutes, etc.
- ; mixer =
-
- ; Software mic volume booster (or attenuator), useful for sound
- ; cards or microphones with poor sensitivity. The volume level
- ; is in dB, ranging from -20.0 to +20.0
- ; boost = n ; mic volume boost in dB
-
- ; Set the callerid for outgoing calls
- ; callerid = John Doe <555-1234>
-
- ; autoanswer = no ; no autoanswer on call
- ; autohangup = yes ; hangup when other party closes
- ; extension = s ; default extension to call
- ; context = default ; default context for outgoing calls
- ; language = "" ; default language
-
- ; If you set overridecontext to 'yes', then the whole dial string
- ; will be interpreted as an extension, which is extremely useful
- ; to dial SIP, IAX and other extensions which use the '@' character.
- ; The default is 'no' just for backward compatibility, but the
- ; suggestion is to change it.
- ; overridecontext = no ; if 'no', the last @ will start the context
- ; if 'yes' the whole string is an extension.
-
- ; low level device parameters in case you have problems with the
- ; device driver on your operating system. You should not touch these
- ; unless you know what you are doing.
- ; queuesize = 10 ; frames in device driver
- ; frags = 8 ; argument to SETFRAGMENT
-
- ; ----------------------------- JITTER BUFFER CONFIGURATION --------------------------
- ; jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of an
- ; OSS channel. Defaults to "no". An enabled jitterbuffer will
- ; be used only if the sending side can create and the receiving
- ; side can not accept jitter. The OSS channel can't accept jitter,
- ; thus an enabled jitterbuffer on the receive OSS side will always
- ; be used if the sending side can create jitter.
-
- ; jbmaxsize = 200 ; Max length of the jitterbuffer in milliseconds.
-
- ; jbresyncthreshold = 1000 ; Jump in the frame timestamps over which the jitterbuffer is
- ; resynchronized. Useful to improve the quality of the voice, with
- ; big jumps in/broken timestamps, usually sent from exotic devices
- ; and programs. Defaults to 1000.
-
- ; jbimpl = fixed ; Jitterbuffer implementation, used on the receiving side of an OSS
- ; channel. Two implementations are currently available - "fixed"
- ; (with size always equals to jbmax-size) and "adaptive" (with
- ; variable size, actually the new jb of IAX2). Defaults to fixed.
-
- ; jbtargetextra = 40 ; This option only affects the jb when 'jbimpl = adaptive' is set.
- ; The option represents the number of milliseconds by which the new
- ; jitter buffer will pad its size. the default is 40, so without
- ; modification, the new jitter buffer will set its size to the jitter
- ; value plus 40 milliseconds. increasing this value may help if your
- ; network normally has low jitter, but occasionally has spikes.
-
- ; jblog = no ; Enables jitterbuffer frame logging. Defaults to "no".
- ; ----------------------------------------------------------------------------------
-
-; below is an entry for a second console channel
-; [card1]
- ; device = /dev/dsp1 ; alternate device
-
-; Below are the settings to support video. You can include them
-; in your general configuration as [general](+,video)
-; The parameters are all available through the CLI as "console name value"
-; Section names used here are only examples.
-
-[my_video](!) ; you can just include in your config
- videodevice = /dev/video0 ; uses your V4L webcam as video source
- videodevice = X11 ; X11 grabber. Dragging on the local display moves the origin.
- videocodec = h263 ; also h261, h263p, h264, mpeg4, ...
-
- ; video_size is the geometry used by the encoder.
- ; Depending on the codec your choice is restricted.
- video_size = 352x288 ; the format WIDTHxHEIGHT is also ok
- video_size = cif ; sqcif, qcif, cif, qvga, vga, ...
-
- ; You can also set the geometry used for the camera, local display and remote display.
- ; The local window is on the right, the remote window is on the left.
- ; Right clicking with the mouse on a video window increases the size,
- ; center-clicking reduces the size.
- camera_size = cif
- remote_size = cif
- local_size = qcif
-
- bitrate = 60000 ; rate told to ffmpeg.
- fps = 5 ; frames per second from the source.
- ; qmin = 3 ; quantizer value passed to the encoder.
-
-; The keypad is made of an image (in any format supported by SDL_image)
-; and some configuration entries indicating the location and function of buttons.
-; These entries can also be contained in the comment field of the image,
-; which is a lot more convenient to manage.
-; E.g. for jpeg you can write them with wrjpgcom (part of libjpeg).
-; The format to define keys is
-; region = <event> <shape> x0 y0 x1 y1 h
-; where <event> is the event to be generated (a digit, pickup, hangup,...)
-; <shape> is the shape of the region (currently 'rect' and 'circle' are
-; supported, the latter is really an ellipse), x0 y0 x1 y1 are the
-; coordinates of the base of the rectangle or main diameter of the ellipse,
-; (they can be rotated) while h is the height of the rectangle or the other
-; diameter of the ellipse.
-;
-[my_skin](!)
- keypad = /tmp/keypad.jpg
- region = 1 rect 19 18 67 18 28
- region = 2 rect 84 18 133 18 28
- region = 3 rect 152 18 201 18 28
- region = 4 rect 19 60 67 60 28
- region = 5 rect 84 60 133 60 28
- region = 6 rect 152 60 201 60 28
- region = 7 rect 19 103 67 103 28
- region = 8 rect 84 103 133 103 28
- region = 9 rect 152 103 201 103 28
- region = * rect 19 146 67 146 28
- region = 0 rect 84 146 133 146 28
- region = # rect 152 146 201 146 28
- region = pickup rect 229 15 267 15 40
- region = hangup rect 230 66 270 64 40
- region = mute circle 232 141 264 141 33
- region = sendvideo circle 235 185 266 185 33
- region = autoanswer rect 228 212 275 212 50
-
-; another skin with entries for the keypad and a small font
-; to write to the message boards in the skin.
-[skin2](!)
- keypad = /tmp/kpad2.jpg
- keypad_font = /tmp/font.png
-
-; to add video support, uncomment this and remember to install
-; the keypad and keypad_font files to the right place
-; [general](+,my_video,skin2)
diff --git a/configure b/configure
index 735a8e9..4d1da0d 100755
--- a/configure
+++ b/configure
@@ -999,10 +999,6 @@
PGSQL_DIR
PGSQL_INCLUDE
PGSQL_LIB
-PBX_OSS
-OSS_DIR
-OSS_INCLUDE
-OSS_LIB
PBX_OSPTK
OSPTK_DIR
OSPTK_INCLUDE
@@ -1296,7 +1292,6 @@
BUILD_VENDOR
BUILD_CPU
BUILD_PLATFORM
-astcachedir
astvarrundir
astlogdir
astspooldir
@@ -1309,6 +1304,7 @@
astlibdir
astheaderdir
astetcdir
+astcachedir
astsbindir
EGREP
GREP
@@ -1349,6 +1345,7 @@
docdir
oldincludedir
includedir
+runstatedir
localstatedir
sharedstatedir
sysconfdir
@@ -1425,7 +1422,6 @@
with_opus
with_opusfile
with_osptk
-with_oss
with_postgres
with_beanstalk
with_pjproject
@@ -1538,6 +1534,7 @@
sysconfdir='${prefix}/etc'
sharedstatedir='${prefix}/com'
localstatedir='${prefix}/var'
+runstatedir='${localstatedir}/run'
includedir='${prefix}/include'
oldincludedir='/usr/include'
docdir='${datarootdir}/doc/${PACKAGE_TARNAME}'
@@ -1790,6 +1787,15 @@
| -silent | --silent | --silen | --sile | --sil)
silent=yes ;;
+ -runstatedir | --runstatedir | --runstatedi | --runstated \
+ | --runstate | --runstat | --runsta | --runst | --runs \
+ | --run | --ru | --r)
+ ac_prev=runstatedir ;;
+ -runstatedir=* | --runstatedir=* | --runstatedi=* | --runstated=* \
+ | --runstate=* | --runstat=* | --runsta=* | --runst=* | --runs=* \
+ | --run=* | --ru=* | --r=*)
+ runstatedir=$ac_optarg ;;
+
-sbindir | --sbindir | --sbindi | --sbind | --sbin | --sbi | --sb)
ac_prev=sbindir ;;
-sbindir=* | --sbindir=* | --sbindi=* | --sbind=* | --sbin=* \
@@ -1927,7 +1933,7 @@
for ac_var in exec_prefix prefix bindir sbindir libexecdir datarootdir \
datadir sysconfdir sharedstatedir localstatedir includedir \
oldincludedir docdir infodir htmldir dvidir pdfdir psdir \
- libdir localedir mandir
+ libdir localedir mandir runstatedir
do
eval ac_val=\$$ac_var
# Remove trailing slashes.
@@ -2080,6 +2086,7 @@
--sysconfdir=DIR read-only single-machine data [PREFIX/etc]
--sharedstatedir=DIR modifiable architecture-independent data [PREFIX/com]
--localstatedir=DIR modifiable single-machine data [PREFIX/var]
+ --runstatedir=DIR modifiable per-process data [LOCALSTATEDIR/run]
--libdir=DIR object code libraries [EPREFIX/lib]
--includedir=DIR C header files [PREFIX/include]
--oldincludedir=DIR C header files for non-gcc [/usr/include]
@@ -2188,7 +2195,6 @@
--with-opus=PATH use Opus files in PATH
--with-opusfile=PATH use Opusfile files in PATH
--with-osptk=PATH use OSP Toolkit files in PATH
- --with-oss=PATH use Open Sound System files in PATH
--with-postgres=PATH use PostgreSQL files in PATH
--with-beanstalk=PATH use Beanstalk Job Queue files in PATH
--with-pjproject=PATH use PJPROJECT files in PATH
@@ -10848,6 +10854,7 @@
+
MISDN_DESCRIP="mISDN user"
MISDN_OPTION="misdn"
PBX_MISDN=0
@@ -11232,38 +11239,6 @@
- OSS_DESCRIP="Open Sound System"
- OSS_OPTION="oss"
- PBX_OSS=0
-
-# Check whether --with-oss was given.
-if test "${with_oss+set}" = set; then :
- withval=$with_oss;
- case ${withval} in
- n|no)
- USE_OSS=no
- # -1 is a magic value used by menuselect to know that the package
- # was disabled, other than 'not found'
- PBX_OSS=-1
- ;;
- y|ye|yes)
- ac_mandatory_list="${ac_mandatory_list} OSS"
- ;;
- *)
- OSS_DIR="${withval}"
- ac_mandatory_list="${ac_mandatory_list} OSS"
- ;;
- esac
-
-fi
-
-
-
-
-
-
-
-
PGSQL_DESCRIP="PostgreSQL"
PGSQL_OPTION="postgres"
PBX_PGSQL=0
@@ -14688,7 +14663,7 @@
We can't simply define LARGE_OFF_T to be 9223372036854775807,
since some C++ compilers masquerading as C compilers
incorrectly reject 9223372036854775807. */
-#define LARGE_OFF_T (((off_t) 1 << 62) - 1 + ((off_t) 1 << 62))
+#define LARGE_OFF_T ((((off_t) 1 << 31) << 31) - 1 + (((off_t) 1 << 31) << 31))
int off_t_is_large[(LARGE_OFF_T % 2147483629 == 721
&& LARGE_OFF_T % 2147483647 == 1)
? 1 : -1];
@@ -14734,7 +14709,7 @@
We can't simply define LARGE_OFF_T to be 9223372036854775807,
since some C++ compilers masquerading as C compilers
incorrectly reject 9223372036854775807. */
-#define LARGE_OFF_T (((off_t) 1 << 62) - 1 + ((off_t) 1 << 62))
+#define LARGE_OFF_T ((((off_t) 1 << 31) << 31) - 1 + (((off_t) 1 << 31) << 31))
int off_t_is_large[(LARGE_OFF_T % 2147483629 == 721
&& LARGE_OFF_T % 2147483647 == 1)
? 1 : -1];
@@ -14758,7 +14733,7 @@
We can't simply define LARGE_OFF_T to be 9223372036854775807,
since some C++ compilers masquerading as C compilers
incorrectly reject 9223372036854775807. */
-#define LARGE_OFF_T (((off_t) 1 << 62) - 1 + ((off_t) 1 << 62))
+#define LARGE_OFF_T ((((off_t) 1 << 31) << 31) - 1 + (((off_t) 1 << 31) << 31))
int off_t_is_large[(LARGE_OFF_T % 2147483629 == 721
&& LARGE_OFF_T % 2147483647 == 1)
? 1 : -1];
@@ -14803,7 +14778,7 @@
We can't simply define LARGE_OFF_T to be 9223372036854775807,
since some C++ compilers masquerading as C compilers
incorrectly reject 9223372036854775807. */
-#define LARGE_OFF_T (((off_t) 1 << 62) - 1 + ((off_t) 1 << 62))
+#define LARGE_OFF_T ((((off_t) 1 << 31) << 31) - 1 + (((off_t) 1 << 31) << 31))
int off_t_is_large[(LARGE_OFF_T % 2147483629 == 721
&& LARGE_OFF_T % 2147483647 == 1)
? 1 : -1];
@@ -14827,7 +14802,7 @@
We can't simply define LARGE_OFF_T to be 9223372036854775807,
since some C++ compilers masquerading as C compilers
incorrectly reject 9223372036854775807. */
-#define LARGE_OFF_T (((off_t) 1 << 62) - 1 + ((off_t) 1 << 62))
+#define LARGE_OFF_T ((((off_t) 1 << 31) << 31) - 1 + (((off_t) 1 << 31) << 31))
int off_t_is_large[(LARGE_OFF_T % 2147483629 == 721
&& LARGE_OFF_T % 2147483647 == 1)
? 1 : -1];
@@ -16127,6 +16102,8 @@
if (*(data + i) != *(data3 + i))
return 14;
close (fd);
+ free (data);
+ free (data3);
return 0;
}
_ACEOF
@@ -24215,216 +24192,6 @@
-# possible places for oss definitions
-
-if test "x${PBX_OSS}" != "x1" -a "${USE_OSS}" != "no"; then
- pbxlibdir=""
- # if --with-OSS=DIR has been specified, use it.
- if test "x${OSS_DIR}" != "x"; then
- if test -d ${OSS_DIR}/lib; then
- pbxlibdir="-L${OSS_DIR}/lib"
- else
- pbxlibdir="-L${OSS_DIR}"
- fi
- fi
-
- # empty lib, assume only headers
- AST_OSS_FOUND=yes
-
-
- # now check for the header.
- if test "${AST_OSS_FOUND}" = "yes"; then
- OSS_LIB="${pbxlibdir} -lossaudio "
- # if --with-OSS=DIR has been specified, use it.
- if test "x${OSS_DIR}" != "x"; then
- OSS_INCLUDE="-I${OSS_DIR}/include"
- fi
- OSS_INCLUDE="${OSS_INCLUDE} "
-
- # check for the header
- ast_ext_lib_check_saved_CPPFLAGS="${CPPFLAGS}"
- CPPFLAGS="${CPPFLAGS} ${OSS_INCLUDE}"
- ac_fn_c_check_header_mongrel "$LINENO" "linux/soundcard.h" "ac_cv_header_linux_soundcard_h" "$ac_includes_default"
-if test "x$ac_cv_header_linux_soundcard_h" = xyes; then :
- OSS_HEADER_FOUND=1
-else
- OSS_HEADER_FOUND=0
-fi
-
-
- CPPFLAGS="${ast_ext_lib_check_saved_CPPFLAGS}"
-
- if test "x${OSS_HEADER_FOUND}" = "x0" ; then
- OSS_LIB=""
- OSS_INCLUDE=""
- else
-
- # only checking headers -> no library
- OSS_LIB=""
-
- PBX_OSS=1
- cat >>confdefs.h <<_ACEOF
-#define HAVE_OSS 1
-_ACEOF
-
- fi
- fi
-fi
-
-
-
-if test "x${PBX_OSS}" != "x1" -a "${USE_OSS}" != "no"; then
- pbxlibdir=""
- # if --with-OSS=DIR has been specified, use it.
- if test "x${OSS_DIR}" != "x"; then
- if test -d ${OSS_DIR}/lib; then
- pbxlibdir="-L${OSS_DIR}/lib"
- else
- pbxlibdir="-L${OSS_DIR}"
- fi
- fi
-
- # empty lib, assume only headers
- AST_OSS_FOUND=yes
-
-
- # now check for the header.
- if test "${AST_OSS_FOUND}" = "yes"; then
- OSS_LIB="${pbxlibdir} -lossaudio "
- # if --with-OSS=DIR has been specified, use it.
- if test "x${OSS_DIR}" != "x"; then
- OSS_INCLUDE="-I${OSS_DIR}/include"
- fi
- OSS_INCLUDE="${OSS_INCLUDE} "
-
- # check for the header
- ast_ext_lib_check_saved_CPPFLAGS="${CPPFLAGS}"
- CPPFLAGS="${CPPFLAGS} ${OSS_INCLUDE}"
- ac_fn_c_check_header_mongrel "$LINENO" "sys/soundcard.h" "ac_cv_header_sys_soundcard_h" "$ac_includes_default"
-if test "x$ac_cv_header_sys_soundcard_h" = xyes; then :
- OSS_HEADER_FOUND=1
-else
- OSS_HEADER_FOUND=0
-fi
-
-
- CPPFLAGS="${ast_ext_lib_check_saved_CPPFLAGS}"
-
- if test "x${OSS_HEADER_FOUND}" = "x0" ; then
- OSS_LIB=""
- OSS_INCLUDE=""
- else
-
- # only checking headers -> no library
- OSS_LIB=""
-
- PBX_OSS=1
- cat >>confdefs.h <<_ACEOF
-#define HAVE_OSS 1
-_ACEOF
-
- fi
- fi
-fi
-
-
-
-if test "x${PBX_OSS}" != "x1" -a "${USE_OSS}" != "no"; then
- pbxlibdir=""
- # if --with-OSS=DIR has been specified, use it.
- if test "x${OSS_DIR}" != "x"; then
- if test -d ${OSS_DIR}/lib; then
- pbxlibdir="-L${OSS_DIR}/lib"
- else
- pbxlibdir="-L${OSS_DIR}"
- fi
- fi
-
- ast_ext_lib_check_save_CFLAGS="${CFLAGS}"
- CFLAGS="${CFLAGS} "
- { $as_echo "$as_me:${as_lineno-$LINENO}: checking for oss_ioctl_mixer in -lossaudio" >&5
-$as_echo_n "checking for oss_ioctl_mixer in -lossaudio... " >&6; }
-if ${ac_cv_lib_ossaudio_oss_ioctl_mixer+:} false; then :
- $as_echo_n "(cached) " >&6
-else
- ac_check_lib_save_LIBS=$LIBS
-LIBS="-lossaudio ${pbxlibdir} $LIBS"
-cat confdefs.h - <<_ACEOF >conftest.$ac_ext
-/* end confdefs.h. */
-
-/* Override any GCC internal prototype to avoid an error.
- Use char because int might match the return type of a GCC
- builtin and then its argument prototype would still apply. */
-#ifdef __cplusplus
-extern "C"
-#endif
-char oss_ioctl_mixer ();
-int
-main ()
-{
-return oss_ioctl_mixer ();
- ;
- return 0;
-}
-_ACEOF
-if ac_fn_c_try_link "$LINENO"; then :
- ac_cv_lib_ossaudio_oss_ioctl_mixer=yes
-else
- ac_cv_lib_ossaudio_oss_ioctl_mixer=no
-fi
-rm -f core conftest.err conftest.$ac_objext \
- conftest$ac_exeext conftest.$ac_ext
-LIBS=$ac_check_lib_save_LIBS
-fi
-{ $as_echo "$as_me:${as_lineno-$LINENO}: result: $ac_cv_lib_ossaudio_oss_ioctl_mixer" >&5
-$as_echo "$ac_cv_lib_ossaudio_oss_ioctl_mixer" >&6; }
-if test "x$ac_cv_lib_ossaudio_oss_ioctl_mixer" = xyes; then :
- AST_OSS_FOUND=yes
-else
- AST_OSS_FOUND=no
-fi
-
- CFLAGS="${ast_ext_lib_check_save_CFLAGS}"
-
-
- # now check for the header.
- if test "${AST_OSS_FOUND}" = "yes"; then
- OSS_LIB="${pbxlibdir} -lossaudio "
- # if --with-OSS=DIR has been specified, use it.
- if test "x${OSS_DIR}" != "x"; then
- OSS_INCLUDE="-I${OSS_DIR}/include"
- fi
- OSS_INCLUDE="${OSS_INCLUDE} "
-
- # check for the header
- ast_ext_lib_check_saved_CPPFLAGS="${CPPFLAGS}"
- CPPFLAGS="${CPPFLAGS} ${OSS_INCLUDE}"
- ac_fn_c_check_header_mongrel "$LINENO" "soundcard.h" "ac_cv_header_soundcard_h" "$ac_includes_default"
-if test "x$ac_cv_header_soundcard_h" = xyes; then :
- OSS_HEADER_FOUND=1
-else
- OSS_HEADER_FOUND=0
-fi
-
-
- CPPFLAGS="${ast_ext_lib_check_saved_CPPFLAGS}"
-
- if test "x${OSS_HEADER_FOUND}" = "x0" ; then
- OSS_LIB=""
- OSS_INCLUDE=""
- else
-
- PBX_OSS=1
- cat >>confdefs.h <<_ACEOF
-#define HAVE_OSS 1
-_ACEOF
-
- fi
- fi
-fi
-
-
-
PG_CONFIG=No
if test "${USE_PGSQL}" != "no"; then
if test "x${PGSQL_DIR}" != "x"; then
diff --git a/configure.ac b/configure.ac
index 2260fe6..02f9f9c 100644
--- a/configure.ac
+++ b/configure.ac
@@ -524,7 +524,6 @@
AST_EXT_LIB_SETUP([OPUS], [Opus], [opus])
AST_EXT_LIB_SETUP([OPUSFILE], [Opusfile], [opusfile])
AST_EXT_LIB_SETUP([OSPTK], [OSP Toolkit], [osptk])
-AST_EXT_LIB_SETUP([OSS], [Open Sound System], [oss])
AST_EXT_LIB_SETUP([PGSQL], [PostgreSQL], [postgres])
AST_EXT_LIB_SETUP([BEANSTALK], [Beanstalk Job Queue], [beanstalk])
@@ -2344,11 +2343,6 @@
AST_EXT_LIB_CHECK([BEANSTALK], [beanstalk], [bs_version], [beanstalk.h])
-# possible places for oss definitions
-AST_EXT_LIB_CHECK([OSS], [ossaudio], [], [linux/soundcard.h])
-AST_EXT_LIB_CHECK([OSS], [ossaudio], [], [sys/soundcard.h])
-AST_EXT_LIB_CHECK([OSS], [ossaudio], [oss_ioctl_mixer], [soundcard.h])
-
PG_CONFIG=No
if test "${USE_PGSQL}" != "no"; then
if test "x${PGSQL_DIR}" != "x"; then
diff --git a/doc/UPGRADE-staging/chan_oss_removal.txt b/doc/UPGRADE-staging/chan_oss_removal.txt
new file mode 100644
index 0000000..062f64b
--- /dev/null
+++ b/doc/UPGRADE-staging/chan_oss_removal.txt
@@ -0,0 +1,6 @@
+Subject: chan_oss
+Master-Only: True
+
+This module was deprecated in Asterisk 16
+and is now being removed in accordance with
+the Asterisk Module Deprecation policy.
diff --git a/include/asterisk/autoconfig.h.in b/include/asterisk/autoconfig.h.in
index 2c770d1..c8225e2 100644
--- a/include/asterisk/autoconfig.h.in
+++ b/include/asterisk/autoconfig.h.in
@@ -603,9 +603,6 @@
/* Define this to indicate the ${OSPTK_DESCRIP} library */
#undef HAVE_OSPTK
-/* Define to 1 if you have the Open Sound System library. */
-#undef HAVE_OSS
-
/* Define to 1 if your system defines the file flag O_EVTONLY in fcntl.h */
#undef HAVE_O_EVTONLY
diff --git a/makeopts.in b/makeopts.in
index 27a5bdc..01c0da0 100644
--- a/makeopts.in
+++ b/makeopts.in
@@ -227,11 +227,6 @@
OSPTK_INCLUDE=@OSPTK_INCLUDE@
OSPTK_LIB=@OSPTK_LIB@
-# ossaudio can optionally use ffmpeg, x11, sdl and sdl_image.
-# Because sdl_image in turn depends on sdl, we don't duplicate the include
-OSS_INCLUDE=@OSS_INCLUDE@ @FFMPEG_INCLUDE@ @SDL_INCLUDE@ @X11_INCLUDE@
-OSS_LIB=@OSS_LIB@ @FFMPEG_LIB@ @SDL_LIB@ @SDL_IMAGE_LIB@ @X11_LIB@
-
PGSQL_INCLUDE=@PGSQL_INCLUDE@
PGSQL_LIB=@PGSQL_LIB@
diff --git a/menuselect/configure b/menuselect/configure
index 1d15c1c..9986b4f 100755
--- a/menuselect/configure
+++ b/menuselect/configure
@@ -696,6 +696,7 @@
docdir
oldincludedir
includedir
+runstatedir
localstatedir
sharedstatedir
sysconfdir
@@ -777,6 +778,7 @@
sysconfdir='${prefix}/etc'
sharedstatedir='${prefix}/com'
localstatedir='${prefix}/var'
+runstatedir='${localstatedir}/run'
includedir='${prefix}/include'
oldincludedir='/usr/include'
docdir='${datarootdir}/doc/${PACKAGE}'
@@ -1029,6 +1031,15 @@
| -silent | --silent | --silen | --sile | --sil)
silent=yes ;;
+ -runstatedir | --runstatedir | --runstatedi | --runstated \
+ | --runstate | --runstat | --runsta | --runst | --runs \
+ | --run | --ru | --r)
+ ac_prev=runstatedir ;;
+ -runstatedir=* | --runstatedir=* | --runstatedi=* | --runstated=* \
+ | --runstate=* | --runstat=* | --runsta=* | --runst=* | --runs=* \
+ | --run=* | --ru=* | --r=*)
+ runstatedir=$ac_optarg ;;
+
-sbindir | --sbindir | --sbindi | --sbind | --sbin | --sbi | --sb)
ac_prev=sbindir ;;
-sbindir=* | --sbindir=* | --sbindi=* | --sbind=* | --sbin=* \
@@ -1166,7 +1177,7 @@
for ac_var in exec_prefix prefix bindir sbindir libexecdir datarootdir \
datadir sysconfdir sharedstatedir localstatedir includedir \
oldincludedir docdir infodir htmldir dvidir pdfdir psdir \
- libdir localedir mandir
+ libdir localedir mandir runstatedir
do
eval ac_val=\$$ac_var
# Remove trailing slashes.
@@ -1319,6 +1330,7 @@
--sysconfdir=DIR read-only single-machine data [PREFIX/etc]
--sharedstatedir=DIR modifiable architecture-independent data [PREFIX/com]
--localstatedir=DIR modifiable single-machine data [PREFIX/var]
+ --runstatedir=DIR modifiable per-process data [LOCALSTATEDIR/run]
--libdir=DIR object code libraries [EPREFIX/lib]
--includedir=DIR C header files [PREFIX/include]
--oldincludedir=DIR C header files for non-gcc [/usr/include]
--
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Gerrit-Project: asterisk
Gerrit-Branch: master
Gerrit-Change-Id: Ib53a42ad974c63871344b95078c61c188e43da99
Gerrit-Change-Number: 16311
Gerrit-PatchSet: 3
Gerrit-Owner: Joshua Colp <jcolp at sangoma.com>
Gerrit-Reviewer: Friendly Automation
Gerrit-Reviewer: George Joseph <gjoseph at digium.com>
Gerrit-Reviewer: Joshua Colp <jcolp at sangoma.com>
Gerrit-Reviewer: Kevin Harwell <kharwell at digium.com>
Gerrit-Reviewer: Sean Bright <sean at seanbright.com>
Gerrit-MessageType: merged
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