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authorMaxim Sobolev <sobomax@FreeBSD.org>2005-05-03 13:39:48 +0000
committerMaxim Sobolev <sobomax@FreeBSD.org>2005-05-03 13:39:48 +0000
commit4382a63faf11cbd8183330f8ce2561b8add4f416 (patch)
tree5bfdc890761336c2db63529e61c1f9c07a451e35 /net/asterisk-bristuff/files/patch-channels::chan_oss.c
parentFix the dd2.cfg existence test in post-install (diff)
pbx_wilcalu.c:
new patch for this file, smooths the effects of an unhandled error Cexiting from poll() and resulting otherwise in this process taking 100% of the CPU rtp.c: updated patch for rtp.c, removes a misleading 'checksum error' message when in reality the recvfrom() just returned no data; chan_oss.c: replacement for the old chan_oss.c - the changes are so massive that having a patch would be completely unreadable. Among other things this lets you change many /dev/dsp parameters from the config file, to ease adapting to the idiosincracies of various sound cards and drivers. It also supports multiple soundcards on the same box, which might be useful in some cases. Submitted by: luigi Add WITHOUT_MYSQL knob. Suggested by: phantom
Notes
Notes: svn path=/head/; revision=134552
Diffstat (limited to 'net/asterisk-bristuff/files/patch-channels::chan_oss.c')
-rw-r--r--net/asterisk-bristuff/files/patch-channels::chan_oss.c1167
1 files changed, 0 insertions, 1167 deletions
diff --git a/net/asterisk-bristuff/files/patch-channels::chan_oss.c b/net/asterisk-bristuff/files/patch-channels::chan_oss.c
deleted file mode 100644
index ef8cfc11d711..000000000000
--- a/net/asterisk-bristuff/files/patch-channels::chan_oss.c
+++ /dev/null
@@ -1,1167 +0,0 @@
-
-$FreeBSD$
-
---- channels/chan_oss.c
-+++ channels/chan_oss.c
-@@ -13,6 +13,8 @@
- *
- * This program is free software, distributed under the terms of
- * the GNU General Public License
-+ *
-+ * FreeBSD changes by Luigi Rizzo, 2005.04.18
- */
-
- #include <asterisk/lock.h>
-@@ -54,21 +56,30 @@
- #endif
-
- /* Lets use 160 sample frames, just like GSM. */
--#define FRAME_SIZE 160
-+/* this corresponds to 20ms of audio. */
-+#define FRAME_SIZE 160 // was 160
-
--/* When you set the frame size, you have to come up with
-- the right buffer format as well. */
-+/*
-+ * When you set the frame size, you have to come up with
-+ * the right buffer format as well.
-+ * OSS lets you define a 'block' size (which should be a power of 2,
-+ * which power is specified in the lower 16 bits) and the number of
-+ * blocks allowed in the buffer (to avoid that the queue grows too large).
-+ * The latter is specified in the top 16 bits.
-+ * We use a block of 64 bytes (0x6), 5 blocks make a frame each sample
-+ * being 2 bytes, and we make room to store two buffers.
-+ * XXX the '10' is magic
-+ */
-+
-+#define N_BLOCKS (buffersize * 5 * 2)
- /* 5 64-byte frames = one frame */
--#define BUFFER_FMT ((buffersize * 10) << 16) | (0x0006);
-+#define BUFFER_FMT (N_BLOCKS << 16) | (0x0006);
-
- /* Don't switch between read/write modes faster than every 300 ms */
--#define MIN_SWITCH_TIME 600
-+#define MIN_SWITCH_TIME 300
-
--static struct timeval lasttime;
-
- static int usecnt;
--static int silencesuppression = 0;
--static int silencethreshold = 1000;
-
-
- AST_MUTEX_DEFINE_STATIC(usecnt_lock);
-@@ -78,16 +89,15 @@
- static char *tdesc = "OSS Console Channel Driver";
- static char *config = "oss.conf";
-
--static char context[AST_MAX_EXTENSION] = "default";
-+static char default_context[AST_MAX_EXTENSION] = "default";
- static char language[MAX_LANGUAGE] = "";
--static char exten[AST_MAX_EXTENSION] = "s";
-+static char oss_exten[AST_MAX_EXTENSION] = "s";
-
--static int hookstate=0;
-
--static short silence[FRAME_SIZE] = {0, };
-
- struct sound {
- int ind;
-+ char *desc;
- short *data;
- int datalen;
- int samplen;
-@@ -96,136 +106,178 @@
- };
-
- static struct sound sounds[] = {
-- { AST_CONTROL_RINGING, ringtone, sizeof(ringtone)/2, 16000, 32000, 1 },
-- { AST_CONTROL_BUSY, busy, sizeof(busy)/2, 4000, 4000, 1 },
-- { AST_CONTROL_CONGESTION, busy, sizeof(busy)/2, 2000, 2000, 1 },
-- { AST_CONTROL_RING, ring10, sizeof(ring10)/2, 16000, 32000, 1 },
-- { AST_CONTROL_ANSWER, answer, sizeof(answer)/2, 2200, 0, 0 },
-+ { AST_CONTROL_RINGING, "RINGING", ringtone, sizeof(ringtone)/2, 16000, 32000, 1 },
-+ { AST_CONTROL_BUSY, "BUSY", busy, sizeof(busy)/2, 4000, 4000, 1 },
-+ { AST_CONTROL_CONGESTION, "CONGESTION", busy, sizeof(busy)/2, 2000, 2000, 1 },
-+ { AST_CONTROL_RING, "RING10", ring10, sizeof(ring10)/2, 16000, 32000, 1 },
-+ { AST_CONTROL_ANSWER, "ANSWER", answer, sizeof(answer)/2, 2200, 0, 0 },
-+ { -1, NULL, 0, 0, 0, 0 }, /* end marker */
- };
-
--/* Sound command pipe */
--static int sndcmd[2];
-+
-
- static struct chan_oss_pvt {
- /* We only have one OSS structure -- near sighted perhaps, but it
- keeps this driver as simple as possible -- as it should be. */
-+ /*
-+ * cursound indicates which in struct sound we play. -1 means nothing,
-+ * any other value is a valid sound, in which case sampsent indicates
-+ * the next sample to send in [0..samplen + silencelen]
-+ * nosound is set to disable the audio data from the channel
-+ * (so we can play the tones etc.).
-+ */
-+ int sndcmd[2]; /* Sound command pipe */
-+ int cursound; /* index of sound to send */
-+ int sampsent; /* # of sound samples sent */
-+ int nosound;
-+
-+ int total_blocks; /* total blocks in the output device */
-+ int sounddev;
-+ enum { M_UNSET, M_FULL, M_READ, M_WRITE } duplex;
-+ int autoanswer;
-+ int autohangup;
-+ int hookstate;
-+ struct timeval lasttime; /* last setformat */
-+
-+ int silencesuppression;
-+ int silencethreshold;
-+ char device[64]; /* device to open */
-+
-+ pthread_t sthread;
-+
- struct ast_channel *owner;
- char exten[AST_MAX_EXTENSION];
- char context[AST_MAX_EXTENSION];
--} oss;
-+} oss = {
-+ .cursound = -1,
-+ .sounddev = -1,
-+ .duplex = M_UNSET, /* XXX check this */
-+ .autoanswer = 1,
-+ .autohangup = 1,
-+ .silencethreshold = 1000,
-+};
-
--static int time_has_passed(void)
-+/*
-+ * returns true if too early to switch
-+ */
-+static int too_early(struct chan_oss_pvt *o)
- {
- struct timeval tv;
- int ms;
- gettimeofday(&tv, NULL);
-- ms = (tv.tv_sec - lasttime.tv_sec) * 1000 +
-- (tv.tv_usec - lasttime.tv_usec) / 1000;
-- if (ms > MIN_SWITCH_TIME)
-+ ms = (tv.tv_sec - o->lasttime.tv_sec) * 1000 +
-+ (tv.tv_usec - o->lasttime.tv_usec) / 1000;
-+ if (ms < MIN_SWITCH_TIME)
- return -1;
- return 0;
- }
-
--/* Number of buffers... Each is FRAMESIZE/8 ms long. For example
-- with 160 sample frames, and a buffer size of 3, we have a 60ms buffer,
-- usually plenty. */
--
--static pthread_t sthread;
--
--#define MAX_BUFFER_SIZE 100
--static int buffersize = 3;
--
--static int full_duplex = 0;
--
--/* Are we reading or writing (simulated full duplex) */
--static int readmode = 1;
--
--/* File descriptor for sound device */
--static int sounddev = -1;
--
--static int autoanswer = 1;
--
--#if 0
--static int calc_loudness(short *frame)
-+/*
-+ * Returns the number of blocks used in the audio output channel
-+ */
-+static int
-+used_blocks(struct chan_oss_pvt *o)
- {
-- int sum = 0;
-- int x;
-- for (x=0;x<FRAME_SIZE;x++) {
-- if (frame[x] < 0)
-- sum -= frame[x];
-- else
-- sum += frame[x];
-+ struct audio_buf_info info;
-+ static int warned=0;
-+ if (ioctl(o->sounddev, SNDCTL_DSP_GETOSPACE, &info)) {
-+ if (!warned) {
-+ ast_log(LOG_WARNING, "Error reading output space\n");
-+ warned++;
- }
-- sum = sum/FRAME_SIZE;
-- return sum;
-+ return 1;
-+ }
-+ if (o->total_blocks == 0) {
-+ 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;
- }
--#endif
-
--static int cursound = -1;
--static int sampsent = 0;
--static int silencelen=0;
--static int offset=0;
--static int nosound=0;
-+static int soundcard_writeframe(struct chan_oss_pvt *o, short *data)
-+{
-+ /* Write an exactly FRAME_SIZE sized of frame */
-+ int res;
-+ static int errors = 0;
-
--static int send_sound(void)
-+ /*
-+ * nothing spectacular.
-+ * If the buffer is full just drop the extra, otherwise write
-+ */
-+ res = used_blocks(o);
-+ if (res > 10) { /* no room to write a block */
-+ errors ++;
-+ if (errors == 0)
-+ ast_log(LOG_WARNING, "write: used %d blocks (%d)\n", res, errors);
-+ return 0;
-+ }
-+ errors = 0;
-+ res = write(o->sounddev, ((void *)data), FRAME_SIZE * 2);
-+ return res;
-+}
-+
-+/*
-+ * handler for 'sound writable' events from the sound thread.
-+ * Builds a frame from the high level description of the sounds,
-+ * (tone+silence) and passes it to the audio device.
-+ */
-+static int send_sound(struct chan_oss_pvt *o)
- {
- short myframe[FRAME_SIZE];
-- int total = FRAME_SIZE;
-- short *frame = NULL;
-- int amt=0;
-- int res;
-- int myoff;
-- audio_buf_info abi;
-- if (cursound > -1) {
-- res = ioctl(sounddev, SNDCTL_DSP_GETOSPACE ,&abi);
-- if (res) {
-- ast_log(LOG_WARNING, "Unable to read output space\n");
-- return -1;
-- }
-- /* Calculate how many samples we can send, max */
-- if (total > (abi.fragments * abi.fragsize / 2))
-- total = abi.fragments * abi.fragsize / 2;
-- res = total;
-- if (sampsent < sounds[cursound].samplen) {
-- myoff=0;
-- while(total) {
-- amt = total;
-- if (amt > (sounds[cursound].datalen - offset))
-- amt = sounds[cursound].datalen - offset;
-- memcpy(myframe + myoff, sounds[cursound].data + offset, amt * 2);
-- total -= amt;
-- offset += amt;
-- sampsent += amt;
-- myoff += amt;
-- if (offset >= sounds[cursound].datalen)
-- offset = 0;
-- }
-- /* Set it up for silence */
-- if (sampsent >= sounds[cursound].samplen)
-- silencelen = sounds[cursound].silencelen;
-- frame = myframe;
-- } else {
-- if (silencelen > 0) {
-- frame = silence;
-- silencelen -= res;
-- } else {
-- if (sounds[cursound].repeat) {
-- /* Start over */
-- sampsent = 0;
-- offset = 0;
-- } else {
-- cursound = -1;
-- nosound = 0;
-- }
-- }
-+ int ofs = 0;
-+ int l_sampsent = o->sampsent;
-+ int l;
-+ struct sound *s;
-+
-+ if (o->cursound < 0) /* no sound to send */
-+ return 0;
-+ s = &sounds[o->cursound];
-+ /*
-+ * prepare a frame
-+ */
-+
-+ for (ofs = 0; ofs < FRAME_SIZE; ofs += l) {
-+ /* take chunks of sound and data until the buffer is full */
-+ l = s->samplen - l_sampsent; /* sound available */
-+ if (l > 0) {
-+ if (l > FRAME_SIZE - ofs)
-+ l = FRAME_SIZE - ofs;
-+ if (0)
-+ ast_log(LOG_WARNING, "send_sound sound %d/%d of %d into %d\n",
-+ l_sampsent, l, s->samplen, ofs);
-+ bcopy(s->data + l_sampsent, myframe + ofs, l*2);
-+ l_sampsent += l;
-+ } else { /* no sound, maybe some silence */
-+ static short silence[FRAME_SIZE] = {0, };
-+
-+ l += s->silencelen;
-+ if (l > 0) {
-+ if (l > FRAME_SIZE - ofs)
-+ l = FRAME_SIZE - ofs;
-+ if (0)
-+ ast_log(LOG_WARNING, "send_sound silence %d/%d of %d into %d\n",
-+ l_sampsent - s->samplen, l, s->silencelen, ofs);
-+ bcopy(silence, myframe + ofs, l*2);
-+ l_sampsent += l;
-+ } else { /* silence is over, restart sound if loop */
-+ if (s->repeat == 0) { /* last block */
-+ ast_log(LOG_WARNING, "send_sound last block\n");
-+ o->cursound = -1;
-+ o->nosound = 0; /* allow audio data */
-+ if (ofs < FRAME_SIZE) /* pad with silence */
-+ bcopy(silence, myframe + ofs, (FRAME_SIZE - ofs)*2);
-+ }
-+ l_sampsent = 0;
- }
-- if (frame)
-- res = write(sounddev, frame, res * 2);
-- if (res > 0)
-- return 0;
-- return res;
-+ }
- }
-- return 0;
-+ l = soundcard_writeframe(o, myframe);
-+ if (l > 0)
-+ o->sampsent = l_sampsent; /* update status */
-+ return 0; /* fake success */
- }
-
- static void *sound_thread(void *unused)
-@@ -235,41 +287,53 @@
- int max;
- int res;
- char ign[4096];
-- if (read(sounddev, ign, sizeof(sounddev)) < 0)
-+ if (read(oss.sounddev, ign, sizeof(ign)) < 0)
- ast_log(LOG_WARNING, "Read error on sound device: %s\n", strerror(errno));
- for(;;) {
- FD_ZERO(&rfds);
- FD_ZERO(&wfds);
-- max = sndcmd[0];
-- FD_SET(sndcmd[0], &rfds);
-+ max = oss.sndcmd[0];
-+ FD_SET(oss.sndcmd[0], &rfds);
- if (!oss.owner) {
-- FD_SET(sounddev, &rfds);
-- if (sounddev > max)
-- max = sounddev;
-+ FD_SET(oss.sounddev, &rfds);
-+ if (oss.sounddev > max)
-+ max = oss.sounddev;
- }
-- if (cursound > -1) {
-- FD_SET(sounddev, &wfds);
-- if (sounddev > max)
-- max = sounddev;
-+ if (oss.cursound > -1) {
-+ FD_SET(oss.sounddev, &wfds);
-+ if (oss.sounddev > max)
-+ max = oss.sounddev;
- }
- res = ast_select(max + 1, &rfds, &wfds, NULL, NULL);
- if (res < 1) {
- ast_log(LOG_WARNING, "select failed: %s\n", strerror(errno));
- continue;
- }
-- if (FD_ISSET(sndcmd[0], &rfds)) {
-- read(sndcmd[0], &cursound, sizeof(cursound));
-- silencelen = 0;
-- offset = 0;
-- sampsent = 0;
-+ if (FD_ISSET(oss.sndcmd[0], &rfds)) { /* read which sound to play from the pipe */
-+ int i, what;
-+
-+ read(oss.sndcmd[0], &what, sizeof(what));
-+ for (i = 0; sounds[i].ind != -1; i++)
-+ if (sounds[i].ind == what) {
-+ oss.cursound = i;
-+ oss.sampsent = 0;
-+ oss.nosound = 1; /* block other audio */
-+ ast_log(LOG_WARNING, "play %s\n", sounds[i].desc);
-+ break;
-+ }
-+ if (sounds[i].ind == -1)
-+ oss.cursound = -1;
-+ ast_log(LOG_WARNING, "cursound %d samplen %d silencelen %d\n",
-+ oss.cursound, oss.cursound >=0 ? sounds[oss.cursound].samplen : -1,
-+ oss.cursound >=0 ? sounds[oss.cursound].silencelen : -1);
- }
-- if (FD_ISSET(sounddev, &rfds)) {
-+ if (FD_ISSET(oss.sounddev, &rfds)) {
- /* Ignore read */
-- if (read(sounddev, ign, sizeof(ign)) < 0)
-+ if (read(oss.sounddev, ign, sizeof(ign)) < 0)
- ast_log(LOG_WARNING, "Read error on sound device: %s\n", strerror(errno));
- }
-- if (FD_ISSET(sounddev, &wfds))
-- if (send_sound())
-+ if (FD_ISSET(oss.sounddev, &wfds))
-+ if (send_sound(&oss) < 0)
- ast_log(LOG_WARNING, "Failed to write sound\n");
- }
- /* Never reached */
-@@ -277,6 +341,20 @@
- }
-
- #if 0
-+static int calc_loudness(short *frame)
-+{
-+ int sum = 0;
-+ int x;
-+ for (x=0;x<FRAME_SIZE;x++) {
-+ if (frame[x] < 0)
-+ sum -= frame[x];
-+ else
-+ sum += frame[x];
-+ }
-+ sum = sum/FRAME_SIZE;
-+ return sum;
-+}
-+
- static int silence_suppress(short *buf)
- {
- #define SILBUF 3
-@@ -284,7 +362,7 @@
- static int silentframes = 0;
- static char silbuf[FRAME_SIZE * 2 * SILBUF];
- static int silbufcnt=0;
-- if (!silencesuppression)
-+ if (!oss.silencesuppression)
- return 0;
- loudness = calc_loudness((short *)(buf));
- if (option_debug)
-@@ -309,7 +387,7 @@
- /* Write any buffered silence we have, it may have something
- important */
- if (silbufcnt) {
-- write(sounddev, silbuf, silbufcnt * FRAME_SIZE);
-+ write(oss.sounddev, silbuf, silbufcnt * FRAME_SIZE);
- silbufcnt = 0;
- }
- }
-@@ -317,27 +395,55 @@
- }
- #endif
-
--static int setformat(void)
-+/*
-+ * 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 = sounddev;
-+ int fmt, desired, res, fd;
- static int warnedalready = 0;
- static int warnedalready2 = 0;
-+
-+ if (o->sounddev >= 0) {
-+ ioctl(o->sounddev, SNDCTL_DSP_RESET, 0);
-+ close(o->sounddev);
-+ o->duplex = M_UNSET;
-+ }
-+ fd = o->sounddev = open(o->device, mode |O_NONBLOCK);
-+ if (o->sounddev < 0) {
-+ ast_log(LOG_WARNING, "Unable to re-open DSP device: %s\n",
-+ strerror(errno));
-+ return -1;
-+ }
-+
-+ gettimeofday(&o->lasttime, NULL);
- fmt = AFMT_S16_LE;
- 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;
- }
-- res = ioctl(fd, SNDCTL_DSP_SETDUPLEX, 0);
--
-- /* Check to see if duplex set (FreeBSD Bug)*/
-- res = ioctl(fd, SNDCTL_DSP_GETCAPS, &fmt);
--
-- if ((fmt & DSP_CAP_DUPLEX) && !res) {
-- if (option_verbose > 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)) {
-+ if (option_verbose > 1)
- ast_verbose(VERBOSE_PREFIX_2 "Console is full duplex\n");
-- full_duplex = -1;
-+ o->duplex = M_FULL;
-+ };
-+ break;
-+ case O_WRONLY:
-+ o->duplex = M_WRITE;
-+ break;
-+ case O_RDONLY:
-+ o->duplex = M_READ;
-+ break;
- }
-+
- fmt = 0;
- res = ioctl(fd, SNDCTL_DSP_STEREO, &fmt);
- if (res < 0) {
-@@ -348,6 +454,7 @@
- desired = 8000;
- fmt = desired;
- res = ioctl(fd, SNDCTL_DSP_SPEED, &fmt);
-+
- if (res < 0) {
- ast_log(LOG_WARNING, "Failed to set audio device to mono\n");
- return -1;
-@@ -357,89 +464,54 @@
- ast_log(LOG_WARNING, "Requested %d Hz, got %d Hz -- sound may be choppy\n", desired, fmt);
- }
- #if 1
-- fmt = BUFFER_FMT;
-+ /*
-+ * on freebsd, SETFRAGMENT does not work very well on some cards.
-+ * Better leave it out
-+ */
-+
-+ // fmt = BUFFER_FMT;
-+ fmt = 0x8; // 256-bytes fragment
- res = ioctl(fd, SNDCTL_DSP_SETFRAGMENT, &fmt);
- if (res < 0) {
- if (!warnedalready2++)
- ast_log(LOG_WARNING, "Unable to set fragment size -- sound may be choppy\n");
- }
- #endif
-+ /* XXX 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;
- }
-
-+/*
-+ * make sure output mode is available. Returns 0 if done,
-+ * 1 if too early to switch, -1 if error
-+ */
- static int soundcard_setoutput(int force)
- {
-- /* Make sure the soundcard is in output mode. */
-- int fd = sounddev;
-- if (full_duplex || (!readmode && !force))
-- return 0;
-- readmode = 0;
-- if (force || time_has_passed()) {
-- ioctl(sounddev, SNDCTL_DSP_RESET, 0);
-- /* Keep the same fd reserved by closing the sound device and copying stdin at the same
-- time. */
-- /* dup2(0, sound); */
-- close(sounddev);
-- fd = open(DEV_DSP, O_WRONLY |O_NONBLOCK);
-- if (fd < 0) {
-- ast_log(LOG_WARNING, "Unable to re-open DSP device: %s\n", strerror(errno));
-- return -1;
-- }
-- /* dup2 will close the original and make fd be sound */
-- if (dup2(fd, sounddev) < 0) {
-- ast_log(LOG_WARNING, "dup2() failed: %s\n", strerror(errno));
-- return -1;
-- }
-- if (setformat()) {
-- return -1;
-- }
-+ if (oss.duplex == M_FULL || (oss.duplex == M_WRITE && !force))
- return 0;
-- }
-- return 1;
-+ if (!force && too_early(&oss))
-+ return 1;
-+ if (setformat(&oss, O_WRONLY))
-+ return -1;
-+ return 0;
- }
-
-+/*
-+ * make sure input mode is available. Returns 0 if done
-+ * 1 if too early to switch, -1 if error
-+ */
- static int soundcard_setinput(int force)
- {
-- int fd = sounddev;
-- if (full_duplex || (readmode && !force))
-- return 0;
-- readmode = -1;
-- if (force || time_has_passed()) {
-- ioctl(sounddev, SNDCTL_DSP_RESET, 0);
-- close(sounddev);
-- /* dup2(0, sound); */
-- fd = open(DEV_DSP, O_RDONLY | O_NONBLOCK);
-- if (fd < 0) {
-- ast_log(LOG_WARNING, "Unable to re-open DSP device: %s\n", strerror(errno));
-- return -1;
-- }
-- /* dup2 will close the original and make fd be sound */
-- if (dup2(fd, sounddev) < 0) {
-- ast_log(LOG_WARNING, "dup2() failed: %s\n", strerror(errno));
-- return -1;
-- }
-- if (setformat()) {
-- return -1;
-- }
-+ if (oss.duplex == M_FULL || (oss.duplex == M_READ && !force))
- return 0;
-- }
-- return 1;
--}
--
--static int soundcard_init(void)
--{
-- /* Assume it's full duplex for starters */
-- int fd = open(DEV_DSP, O_RDWR | O_NONBLOCK);
-- if (fd < 0) {
-- ast_log(LOG_WARNING, "Unable to open %s: %s\n", DEV_DSP, strerror(errno));
-- return fd;
-- }
-- gettimeofday(&lasttime, NULL);
-- sounddev = fd;
-- setformat();
-- if (!full_duplex)
-- soundcard_setinput(1);
-- return sounddev;
-+ if (!force && too_early(&oss))
-+ return 1;
-+ if (setformat(&oss, O_RDONLY))
-+ return -1;
-+ return 0;
- }
-
- static int oss_digit(struct ast_channel *c, char digit)
-@@ -454,120 +526,81 @@
- return 0;
- }
-
-+/* request to play a sound on the speaker */
-+#define RING(x) { int what = x; write(oss.sndcmd[1], &what, sizeof(what)); }
-+
- static int oss_call(struct ast_channel *c, char *dest, int timeout)
- {
-- int res = 3;
- struct ast_frame f = { 0, };
- ast_verbose( " << Call placed to '%s' on console >> \n", dest);
-- if (autoanswer) {
-+ if (oss.autoanswer) {
- ast_verbose( " << Auto-answered >> \n" );
- f.frametype = AST_FRAME_CONTROL;
- f.subclass = AST_CONTROL_ANSWER;
- ast_queue_frame(c, &f);
- } else {
-- nosound = 1;
- ast_verbose( " << Type 'answer' to answer, or use 'autoanswer' for future calls >> \n");
- f.frametype = AST_FRAME_CONTROL;
- f.subclass = AST_CONTROL_RINGING;
- ast_queue_frame(c, &f);
-- write(sndcmd[1], &res, sizeof(res));
-+ RING(AST_CONTROL_RING);
- }
- return 0;
- }
-
- static void answer_sound(void)
- {
-- int res;
-- nosound = 1;
-- res = 4;
-- write(sndcmd[1], &res, sizeof(res));
--
-+ RING(AST_CONTROL_ANSWER);
- }
-
- static int oss_answer(struct ast_channel *c)
- {
- ast_verbose( " << Console call has been answered >> \n");
-- answer_sound();
-+ answer_sound(); /* XXX do we really need it ? considering we shut down immediately... */
- ast_setstate(c, AST_STATE_UP);
-- cursound = -1;
-- nosound=0;
-+ oss.cursound = -1;
-+ oss.nosound=0;
- return 0;
- }
-
- static int oss_hangup(struct ast_channel *c)
- {
-- int res = 0;
-- cursound = -1;
-+ oss.cursound = -1;
- c->pvt->pvt = NULL;
- oss.owner = NULL;
- ast_verbose( " << Hangup on console >> \n");
- ast_mutex_lock(&usecnt_lock);
- usecnt--;
- ast_mutex_unlock(&usecnt_lock);
-- if (hookstate) {
-- if (autoanswer) {
-+ if (oss.hookstate) {
-+ if (oss.autoanswer || oss.autohangup) {
- /* Assume auto-hangup too */
-- hookstate = 0;
-+ oss.hookstate = 0;
- } else {
- /* Make congestion noise */
-- res = 2;
-- write(sndcmd[1], &res, sizeof(res));
-+ RING(AST_CONTROL_CONGESTION);
- }
- }
- return 0;
- }
-
--static int soundcard_writeframe(short *data)
--{
-- /* Write an exactly FRAME_SIZE sized of frame */
-- static int bufcnt = 0;
-- static short buffer[FRAME_SIZE * MAX_BUFFER_SIZE * 5];
-- struct audio_buf_info info;
-- int res;
-- int fd = sounddev;
-- static int warned=0;
-- if (ioctl(fd, SNDCTL_DSP_GETOSPACE, &info)) {
-- if (!warned)
-- ast_log(LOG_WARNING, "Error reading output space\n");
-- bufcnt = buffersize;
-- warned++;
-- }
-- if ((info.fragments >= buffersize * 5) && (bufcnt == buffersize)) {
-- /* We've run out of stuff, buffer again */
-- bufcnt = 0;
-- }
-- if (bufcnt == buffersize) {
-- /* Write sample immediately */
-- res = write(fd, ((void *)data), FRAME_SIZE * 2);
-- } else {
-- /* Copy the data into our buffer */
-- res = FRAME_SIZE * 2;
-- memcpy(buffer + (bufcnt * FRAME_SIZE), data, FRAME_SIZE * 2);
-- bufcnt++;
-- if (bufcnt == buffersize) {
-- res = write(fd, ((void *)buffer), FRAME_SIZE * 2 * buffersize);
-- }
-- }
-- return res;
--}
--
--
-+/* used for data coming from the network */
- static int oss_write(struct ast_channel *chan, struct ast_frame *f)
- {
- int res;
-- static char sizbuf[8000];
-- static int sizpos = 0;
-- int len = sizpos;
-- int pos;
-+ int src;
-+
-+ // ast_log(LOG_WARNING, "oss_write size %d\n", f->datalen);
- /* Immediately return if no sound is enabled */
-- if (nosound)
-+ if (oss.nosound)
- return 0;
- /* Stop any currently playing sound */
-- cursound = -1;
-- if (!full_duplex) {
-+ oss.cursound = -1;
-+ if (oss.duplex != M_FULL) {
-+ /* XXX check this, looks weird! */
- /* If we're half duplex, we have to switch to read mode
- to honor immediate needs if necessary */
-- res = soundcard_setinput(1);
-+ res = soundcard_setinput(1); /* force set if not full_duplex */
- if (res < 0) {
- ast_log(LOG_WARNING, "Unable to set device to input mode\n");
- return -1;
-@@ -583,21 +616,30 @@
- so just pretend we wrote it */
- return 0;
- }
-- /* We have to digest the frame in 160-byte portions */
-- if (f->datalen > sizeof(sizbuf) - sizpos) {
-- ast_log(LOG_WARNING, "Frame too large\n");
-- return -1;
-- }
-- memcpy(sizbuf + sizpos, f->data, f->datalen);
-- len += f->datalen;
-- pos = 0;
-- while(len - pos > FRAME_SIZE * 2) {
-- soundcard_writeframe((short *)(sizbuf + pos));
-- pos += FRAME_SIZE * 2;
-+ /*
-+ * we could receive a sample which is not a multiple of our FRAME_SIZE,
-+ * so we buffer it locally and write to the device in FRAME_SIZE
-+ * chunks, keeping the residue stored for future use.
-+ */
-+
-+ src = 0; /* read position into f->data */
-+ while ( src < f->datalen ) {
-+ static char buf[FRAME_SIZE*2];
-+ static int dst = 0;
-+ int l = sizeof(buf) - dst; /* how much room in the buffer */
-+
-+ if (f->datalen - src >= l) { /* enough to fill a frame */
-+ memcpy(buf + dst, f->data + src, l);
-+ soundcard_writeframe(&oss, (short *)buf);
-+ src += l;
-+ dst = 0;
-+ } else { /* copy residue */
-+ l = f->datalen - src;
-+ memcpy(buf + dst, f->data + src, l);
-+ src += l; /* but really, we are done */
-+ dst += l;
-+ }
- }
-- if (len - pos)
-- memmove(sizbuf, sizbuf + pos, len - pos);
-- sizpos = len - pos;
- return 0;
- }
-
-@@ -628,18 +670,15 @@
- ast_log(LOG_WARNING, "Unable to set input mode\n");
- return NULL;
- }
-- if (res > 0) {
-+ if (res > 0) { /* too early to switch ? */
- /* Theoretically shouldn't happen, but anyway, return a NULL frame */
- return &f;
- }
-- res = read(sounddev, buf + AST_FRIENDLY_OFFSET + readpos, FRAME_SIZE * 2 - readpos);
-- if (res < 0) {
-- ast_log(LOG_WARNING, "Error reading from sound device (If you're running 'artsd' then kill it): %s\n", strerror(errno));
--#if 0
-- CRASH;
--#endif
-- return NULL;
-- }
-+
-+ res = read(oss.sounddev, buf + AST_FRIENDLY_OFFSET + readpos, FRAME_SIZE * 2 - readpos);
-+ // ast_log(LOG_WARNING, "oss_read() fd %d got %d\n", oss.sounddev, res);
-+ if (res < 0) /* audio data not ready, return a NULL frame */
-+ return &f;
- readpos += res;
-
- if (readpos >= FRAME_SIZE * 2) {
-@@ -682,64 +721,66 @@
- int res;
- switch(cond) {
- case AST_CONTROL_BUSY:
-- res = 1;
-- break;
- case AST_CONTROL_CONGESTION:
-- res = 2;
-- break;
- case AST_CONTROL_RINGING:
-- res = 0;
-+ res = cond;
- break;
- case -1:
-- cursound = -1;
-+ oss.cursound = -1;
- return 0;
- default:
- ast_log(LOG_WARNING, "Don't know how to display condition %d on %s\n", cond, chan->name);
- return -1;
- }
- if (res > -1) {
-- write(sndcmd[1], &res, sizeof(res));
-+ RING(res);
- }
- return 0;
- }
-
--static struct ast_channel *oss_new(struct chan_oss_pvt *p, int state)
-+static struct ast_channel *oss_new(struct chan_oss_pvt *oss, int state)
- {
- struct ast_channel *tmp;
-+ struct ast_channel_pvt *pvt;
-+
- tmp = ast_channel_alloc(1);
-- if (tmp) {
-- snprintf(tmp->name, sizeof(tmp->name), "OSS/%s", DEV_DSP + 5);
-- tmp->type = type;
-- tmp->fds[0] = sounddev;
-- tmp->nativeformats = AST_FORMAT_SLINEAR;
-- tmp->pvt->pvt = p;
-- tmp->pvt->send_digit = oss_digit;
-- tmp->pvt->send_text = oss_text;
-- tmp->pvt->hangup = oss_hangup;
-- tmp->pvt->answer = oss_answer;
-- tmp->pvt->read = oss_read;
-- tmp->pvt->call = oss_call;
-- tmp->pvt->write = oss_write;
-- tmp->pvt->indicate = oss_indicate;
-- tmp->pvt->fixup = oss_fixup;
-- if (strlen(p->context))
-- strncpy(tmp->context, p->context, sizeof(tmp->context)-1);
-- if (strlen(p->exten))
-- strncpy(tmp->exten, p->exten, sizeof(tmp->exten)-1);
-- if (strlen(language))
-- strncpy(tmp->language, language, sizeof(tmp->language)-1);
-- p->owner = tmp;
-- ast_setstate(tmp, state);
-- ast_mutex_lock(&usecnt_lock);
-- usecnt++;
-- ast_mutex_unlock(&usecnt_lock);
-- ast_update_use_count();
-- if (state != AST_STATE_DOWN) {
-- if (ast_pbx_start(tmp)) {
-- ast_log(LOG_WARNING, "Unable to start PBX on %s\n", tmp->name);
-- ast_hangup(tmp);
-- tmp = NULL;
-- }
-+ if (tmp == NULL)
-+ return NULL;
-+ snprintf(tmp->name, sizeof(tmp->name), "OSS/%s", oss->device + 5);
-+ tmp->type = type;
-+ tmp->fds[0] = oss->sounddev;
-+ tmp->nativeformats = AST_FORMAT_SLINEAR;
-+ pvt = tmp->pvt;
-+ pvt->pvt = oss;
-+#if 1
-+ pvt->send_digit = oss_digit;
-+ pvt->send_text = oss_text;
-+ pvt->hangup = oss_hangup;
-+ pvt->answer = oss_answer;
-+ pvt->read = oss_read;
-+ pvt->call = oss_call;
-+ pvt->write = oss_write;
-+ pvt->indicate = oss_indicate;
-+ pvt->fixup = oss_fixup;
-+#endif
-+ if (strlen(oss->context))
-+ strncpy(tmp->context, oss->context, sizeof(tmp->context)-1);
-+ if (strlen(oss->exten))
-+ strncpy(tmp->exten, oss->exten, sizeof(tmp->exten)-1);
-+ if (strlen(language))
-+ strncpy(tmp->language, language, sizeof(tmp->language)-1);
-+ oss->owner = tmp;
-+ ast_setstate(tmp, state);
-+ ast_mutex_lock(&usecnt_lock);
-+ usecnt++;
-+ ast_mutex_unlock(&usecnt_lock);
-+ ast_update_use_count();
-+ if (state != AST_STATE_DOWN) {
-+ if (ast_pbx_start(tmp)) {
-+ ast_log(LOG_WARNING, "Unable to start PBX on %s\n", tmp->name);
-+ ast_hangup(tmp);
-+ tmp = NULL;
-+ /* XXX what about oss->owner and the channel itself ? */
- }
- }
- return tmp;
-@@ -770,13 +811,13 @@
- if ((argc != 1) && (argc != 2))
- return RESULT_SHOWUSAGE;
- if (argc == 1) {
-- ast_cli(fd, "Auto answer is %s.\n", autoanswer ? "on" : "off");
-+ ast_cli(fd, "Auto answer is %s.\n", oss.autoanswer ? "on" : "off");
- return RESULT_SUCCESS;
- } else {
- if (!strcasecmp(argv[1], "on"))
-- autoanswer = -1;
-+ oss.autoanswer = -1;
- else if (!strcasecmp(argv[1], "off"))
-- autoanswer = 0;
-+ oss.autoanswer = 0;
- else
- return RESULT_SHOWUSAGE;
- }
-@@ -788,12 +829,14 @@
- #ifndef MIN
- #define MIN(a,b) ((a) < (b) ? (a) : (b))
- #endif
-+ int l = strlen(word);
-+
- switch(state) {
- case 0:
-- if (strlen(word) && !strncasecmp(word, "on", MIN(strlen(word), 2)))
-+ if (l && !strncasecmp(word, "on", MIN(l, 2)))
- return strdup("on");
- case 1:
-- if (strlen(word) && !strncasecmp(word, "off", MIN(strlen(word), 3)))
-+ if (l && !strncasecmp(word, "off", MIN(l, 3)))
- return strdup("off");
- default:
- return NULL;
-@@ -816,8 +859,8 @@
- ast_cli(fd, "No one is calling us\n");
- return RESULT_FAILURE;
- }
-- hookstate = 1;
-- cursound = -1;
-+ oss.hookstate = 1;
-+ oss.cursound = -1;
- ast_queue_frame(oss.owner, &f);
- answer_sound();
- return RESULT_SUCCESS;
-@@ -863,12 +906,12 @@
- {
- if (argc != 1)
- return RESULT_SHOWUSAGE;
-- cursound = -1;
-- if (!oss.owner && !hookstate) {
-+ oss.cursound = -1;
-+ if (!oss.owner && !oss.hookstate) {
- ast_cli(fd, "No call to hangup up\n");
- return RESULT_FAILURE;
- }
-- hookstate = 0;
-+ oss.hookstate = 0;
- if (oss.owner) {
- ast_queue_hangup(oss.owner);
- }
-@@ -900,8 +943,8 @@
- }
- return RESULT_SUCCESS;
- }
-- mye = exten;
-- myc = context;
-+ mye = oss_exten;
-+ myc = default_context;
- if (argc == 2) {
- char *stringp=NULL;
- strncpy(tmp, argv[1], sizeof(tmp)-1);
-@@ -916,7 +959,7 @@
- if (ast_exists_extension(NULL, myc, mye, 1, NULL)) {
- strncpy(oss.exten, mye, sizeof(oss.exten)-1);
- strncpy(oss.context, myc, sizeof(oss.context)-1);
-- hookstate = 1;
-+ oss.hookstate = 1;
- oss_new(&oss, AST_STATE_RINGING);
- } else
- ast_cli(fd, "No such extension '%s' in context '%s'\n", mye, myc);
-@@ -974,21 +1017,47 @@
- int res;
- int x;
- struct ast_config *cfg;
-- struct ast_variable *v;
-- res = pipe(sndcmd);
-+
-+ res = pipe(oss.sndcmd);
- if (res) {
- ast_log(LOG_ERROR, "Unable to create pipe\n");
- return -1;
- }
-- res = soundcard_init();
-- if (res < 0) {
-+ /* load config file */
-+ if ((cfg = ast_load(config))) {
-+ struct ast_variable *v = ast_variable_browse(cfg, "general");
-+ while(v) {
-+ if (!strcasecmp(v->name, "autoanswer"))
-+ oss.autoanswer = ast_true(v->value);
-+ else if (!strcasecmp(v->name, "autohangup"))
-+ oss.autohangup = ast_true(v->value);
-+ else if (!strcasecmp(v->name, "oss.silencesuppression"))
-+ oss.silencesuppression = ast_true(v->value);
-+ else if (!strcasecmp(v->name, "silencethreshold"))
-+ oss.silencethreshold = atoi(v->value);
-+ else if (!strcasecmp(v->name, "device"))
-+ strncpy(oss.device, v->value, sizeof(oss.device)-1);
-+ else if (!strcasecmp(v->name, "context"))
-+ strncpy(default_context, v->value, sizeof(default_context)-1);
-+ else if (!strcasecmp(v->name, "language"))
-+ strncpy(language, v->value, sizeof(language)-1);
-+ else if (!strcasecmp(v->name, "extension"))
-+ strncpy(oss_exten, v->value, sizeof(oss_exten)-1);
-+ v=v->next;
-+ }
-+ ast_destroy(cfg);
-+ }
-+ if (!strlen(oss.device))
-+ strncpy(oss.device, DEV_DSP, sizeof(oss.device)-1);
-+ if (setformat(&oss, O_RDWR) < 0) { /* open device */
- if (option_verbose > 1) {
- ast_verbose(VERBOSE_PREFIX_2 "No sound card detected -- console channel will be unavailable\n");
- ast_verbose(VERBOSE_PREFIX_2 "Turn off OSS support by adding 'noload=chan_oss.so' in /etc/asterisk/modules.conf\n");
- }
- return 0;
- }
-- if (!full_duplex)
-+ soundcard_setinput(1); /* force set if not full_duplex */
-+ if (oss.duplex != M_FULL)
- ast_log(LOG_WARNING, "XXX I don't work right with non-full duplex sound cards XXX\n");
- res = ast_channel_register(type, tdesc, AST_FORMAT_SLINEAR, oss_request);
- if (res < 0) {
-@@ -997,26 +1066,7 @@
- }
- for (x=0;x<sizeof(myclis)/sizeof(struct ast_cli_entry); x++)
- ast_cli_register(myclis + x);
-- if ((cfg = ast_load(config))) {
-- v = ast_variable_browse(cfg, "general");
-- while(v) {
-- if (!strcasecmp(v->name, "autoanswer"))
-- autoanswer = ast_true(v->value);
-- else if (!strcasecmp(v->name, "silencesuppression"))
-- silencesuppression = ast_true(v->value);
-- else if (!strcasecmp(v->name, "silencethreshold"))
-- silencethreshold = atoi(v->value);
-- else if (!strcasecmp(v->name, "context"))
-- strncpy(context, v->value, sizeof(context)-1);
-- else if (!strcasecmp(v->name, "language"))
-- strncpy(language, v->value, sizeof(language)-1);
-- else if (!strcasecmp(v->name, "extension"))
-- strncpy(exten, v->value, sizeof(exten)-1);
-- v=v->next;
-- }
-- ast_destroy(cfg);
-- }
-- ast_pthread_create(&sthread, NULL, sound_thread, NULL);
-+ ast_pthread_create(&oss.sthread, NULL, sound_thread, NULL);
- return 0;
- }
-
-@@ -1027,15 +1077,16 @@
- int x;
- for (x=0;x<sizeof(myclis)/sizeof(struct ast_cli_entry); x++)
- ast_cli_unregister(myclis + x);
-- close(sounddev);
-- if (sndcmd[0] > 0) {
-- close(sndcmd[0]);
-- close(sndcmd[1]);
-+ close(oss.sounddev);
-+ if (oss.sndcmd[0] > 0) {
-+ close(oss.sndcmd[0]);
-+ close(oss.sndcmd[1]);
- }
- if (oss.owner)
- ast_softhangup(oss.owner, AST_SOFTHANGUP_APPUNLOAD);
- if (oss.owner)
- return -1;
-+ /* XXX what about the thread ? */
- return 0;
- }
-