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https://github.com/asterisk/asterisk.git
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Version 0.1.7 from FTP
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@236 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
157
apps/app_mp3.c
157
apps/app_mp3.c
@@ -59,12 +59,13 @@ static int mp3_exec(struct ast_channel *chan, void *data)
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{
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int res=0;
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struct localuser *u;
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struct ast_channel *trans;
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int fds[2];
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int rfds[2];
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int ms = -1;
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int pid;
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int us;
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int exception;
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int owriteformat;
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struct timeval tv;
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struct timeval last;
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struct ast_frame *f;
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@@ -85,87 +86,90 @@ static int mp3_exec(struct ast_channel *chan, void *data)
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}
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LOCAL_USER_ADD(u);
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ast_stopstream(chan);
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if (chan->format & AST_FORMAT_SLINEAR)
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trans = chan;
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else
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trans = ast_translator_create(chan, AST_FORMAT_SLINEAR, AST_DIRECTION_OUT);
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if (trans) {
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res = mp3play((char *)data, fds[1]);
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if (res >= 0) {
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pid = res;
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/* Order is important -- there's almost always going to be mp3... we want to prioritize the
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user */
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rfds[0] = trans->fd;
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rfds[1] = fds[0];
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for (;;) {
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CHECK_BLOCKING(trans);
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res = ast_waitfor_n_fd(rfds, 2, &ms);
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trans->blocking = 0;
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if (res < 1) {
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ast_log(LOG_DEBUG, "Hangup detected\n");
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res = -1;
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break;
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} else if (res == trans->fd) {
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f = ast_read(trans);
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if (!f) {
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ast_log(LOG_DEBUG, "Null frame == hangup() detected\n");
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res = -1;
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break;
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}
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if (f->frametype == AST_FRAME_DTMF) {
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ast_log(LOG_DEBUG, "User pressed a key\n");
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ast_frfree(f);
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res = 0;
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break;
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}
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ast_frfree(f);
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} else if (res == fds[0]) {
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gettimeofday(&tv, NULL);
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if (last.tv_sec || last.tv_usec) {
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/* We should wait at least a frame length */
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us = sizeof(myf.frdata) / 16 * 1000;
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/* Subtract 1,000,000 us for each second late we've passed */
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us -= (tv.tv_sec - last.tv_sec) * 1000000;
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/* And one for each us late we've passed */
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us -= (tv.tv_usec - last.tv_usec);
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/* Sleep that long if needed */
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if (us > 0)
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usleep(us);
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}
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last = tv;
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res = read(fds[0], myf.frdata, sizeof(myf.frdata));
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if (res > 0) {
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myf.f.frametype = AST_FRAME_VOICE;
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myf.f.subclass = AST_FORMAT_SLINEAR;
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myf.f.datalen = res;
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myf.f.timelen = res / 16;
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myf.f.mallocd = 0;
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myf.f.offset = AST_FRIENDLY_OFFSET;
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myf.f.src = __PRETTY_FUNCTION__;
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myf.f.data = myf.frdata;
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if (ast_write(trans, &myf.f) < 0) {
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res = -1;
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break;
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}
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} else {
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ast_log(LOG_DEBUG, "No more mp3\n");
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res = 0;
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}
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} else {
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ast_log(LOG_DEBUG, "HuhHHH?\n");
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owriteformat = chan->writeformat;
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res = ast_set_write_format(chan, AST_FORMAT_SLINEAR);
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if (res < 0) {
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ast_log(LOG_WARNING, "Unable to set write format to signed linear\n");
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return -1;
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}
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res = mp3play((char *)data, fds[1]);
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if (res >= 0) {
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pid = res;
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/* Order is important -- there's almost always going to be mp3... we want to prioritize the
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user */
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rfds[0] = chan->fd;
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rfds[1] = fds[0];
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for (;;) {
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CHECK_BLOCKING(chan);
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res = ast_waitfor_n_fd(rfds, 2, &ms, &exception);
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chan->blocking = 0;
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if (res < 1) {
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ast_log(LOG_DEBUG, "Hangup detected\n");
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res = -1;
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break;
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} else if (res == chan->fd) {
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if (exception)
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chan->exception = 1;
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f = ast_read(chan);
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if (!f) {
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ast_log(LOG_DEBUG, "Null frame == hangup() detected\n");
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res = -1;
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break;
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}
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if (f->frametype == AST_FRAME_DTMF) {
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ast_log(LOG_DEBUG, "User pressed a key\n");
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ast_frfree(f);
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res = 0;
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break;
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}
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ast_frfree(f);
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} else if (res == fds[0]) {
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gettimeofday(&tv, NULL);
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if (last.tv_sec || last.tv_usec) {
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/* We should wait at least a frame length */
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us = sizeof(myf.frdata) / 16 * 1000;
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/* Subtract 1,000,000 us for each second late we've passed */
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us -= (tv.tv_sec - last.tv_sec) * 1000000;
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/* And one for each us late we've passed */
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us -= (tv.tv_usec - last.tv_usec);
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/* Sleep that long if needed */
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if (us > 0)
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usleep(us);
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}
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last = tv;
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res = read(fds[0], myf.frdata, sizeof(myf.frdata));
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if (res > 0) {
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myf.f.frametype = AST_FRAME_VOICE;
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myf.f.subclass = AST_FORMAT_SLINEAR;
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myf.f.datalen = res;
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myf.f.timelen = res / 16;
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myf.f.mallocd = 0;
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myf.f.offset = AST_FRIENDLY_OFFSET;
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myf.f.src = __PRETTY_FUNCTION__;
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myf.f.data = myf.frdata;
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if (ast_write(chan, &myf.f) < 0) {
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res = -1;
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break;
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}
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} else {
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ast_log(LOG_DEBUG, "No more mp3\n");
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res = 0;
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}
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} else {
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ast_log(LOG_DEBUG, "HuhHHH?\n");
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res = -1;
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break;
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}
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kill(pid, SIGTERM);
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}
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if (trans != chan)
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ast_translator_destroy(trans);
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} else
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ast_log(LOG_WARNING, "No translator channel available\n");
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kill(pid, SIGTERM);
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}
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close(fds[0]);
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close(fds[1]);
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LOCAL_USER_REMOVE(u);
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if (!res)
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ast_set_write_format(chan, owriteformat);
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return res;
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}
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@@ -191,3 +195,8 @@ int usecount(void)
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STANDARD_USECOUNT(res);
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return res;
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}
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char *key()
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{
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return ASTERISK_GPL_KEY;
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}
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