/*! \brief Helper function which gets the format for a Snoop channel based on the channel being snooped on */
static void snoop_determine_format(struct ast_channel *chan, struct stasis_app_snoop *snoop)
{
SCOPED_CHANNELLOCK(lock, chan);
unsigned int rate = MAX(ast_format_get_sample_rate(ast_channel_rawwriteformat(chan)),
ast_format_get_sample_rate(ast_channel_rawreadformat(chan)));
snoop->spy_format = ast_format_cache_get_slin_by_rate(rate);
}
static int resamp_new(struct ast_trans_pvt *pvt)
{
int err;
if (!(pvt->pvt = speex_resampler_init(1, pvt->t->src_codec.sample_rate, pvt->t->dst_codec.sample_rate, 5, &err))) {
return -1;
}
ast_assert(pvt->f.subclass.format == NULL);
pvt->f.subclass.format = ao2_bump(ast_format_cache_get_slin_by_rate(pvt->t->dst_codec.sample_rate));
return 0;
}
static int mp3_exec(struct ast_channel *chan, const char *data)
{
int res=0;
int fds[2];
int ms = -1;
int pid = -1;
RAII_VAR(struct ast_format *, owriteformat, NULL, ao2_cleanup);
int timeout = 2000;
struct timeval next;
struct ast_frame *f;
struct myframe {
struct ast_frame f;
char offset[AST_FRIENDLY_OFFSET];
short frdata[160];
} myf = {
.f = { 0, },
};
struct ast_format * native_format;
unsigned int sampling_rate;
struct ast_format * write_format;
if (ast_strlen_zero(data)) {
ast_log(LOG_WARNING, "MP3 Playback requires an argument (filename)\n");
return -1;
}
if (pipe(fds)) {
ast_log(LOG_WARNING, "Unable to create pipe\n");
return -1;
}
ast_stopstream(chan);
native_format = ast_format_cap_get_format(ast_channel_nativeformats(chan), 0);
sampling_rate = ast_format_get_sample_rate(native_format);
write_format = ast_format_cache_get_slin_by_rate(sampling_rate);
owriteformat = ao2_bump(ast_channel_writeformat(chan));
res = ast_set_write_format(chan, write_format);
if (res < 0) {
ast_log(LOG_WARNING, "Unable to set write format to signed linear\n");
return -1;
}
myf.f.frametype = AST_FRAME_VOICE;
myf.f.subclass.format = write_format;
myf.f.mallocd = 0;
myf.f.offset = AST_FRIENDLY_OFFSET;
myf.f.src = __PRETTY_FUNCTION__;
myf.f.delivery.tv_sec = 0;
myf.f.delivery.tv_usec = 0;
myf.f.data.ptr = myf.frdata;
res = mp3play(data, sampling_rate, fds[1]);
if (!strncasecmp(data, "http://", 7)) {
timeout = 10000;
}
/* Wait 1000 ms first */
next = ast_tvnow();
next.tv_sec += 1;
if (res >= 0) {
pid = res;
/* Order is important -- there's almost always going to be mp3... we want to prioritize the
user */
for (;;) {
ms = ast_tvdiff_ms(next, ast_tvnow());
if (ms <= 0) {
res = timed_read(fds[0], myf.frdata, sizeof(myf.frdata), timeout);
if (res > 0) {
myf.f.datalen = res;
myf.f.samples = res / 2;
if (ast_write(chan, &myf.f) < 0) {
res = -1;
break;
}
} else {
ast_debug(1, "No more mp3\n");
res = 0;
break;
}
next = ast_tvadd(next, ast_samp2tv(myf.f.samples, sampling_rate));
} else {
ms = ast_waitfor(chan, ms);
if (ms < 0) {
ast_debug(1, "Hangup detected\n");
res = -1;
break;
}
if (ms) {
f = ast_read(chan);
if (!f) {
ast_debug(1, "Null frame == hangup() detected\n");
res = -1;
break;
}
if (f->frametype == AST_FRAME_DTMF) {
ast_debug(1, "User pressed a key\n");
ast_frfree(f);
res = 0;
break;
}
ast_frfree(f);
}
}
}
}
close(fds[0]);
close(fds[1]);
if (pid > -1)
kill(pid, SIGKILL);
if (!res && owriteformat)
ast_set_write_format(chan, owriteformat);
ast_frfree(&myf.f);
return res;
}
