AudioEngine::instance()->Stopped.connect (forever_connections, MISSING_INVALIDATOR, boost::bind (&ARDOUR_UI::engine_stopped, this), gui_context());
AudioEngine::instance()->SampleRateChanged.connect (forever_connections, MISSING_INVALIDATOR, boost::bind (&ARDOUR_UI::update_sample_rate, this, _1), gui_context());
+ AudioEngine::instance()->BufferSizeChanged.connect (forever_connections, MISSING_INVALIDATOR, boost::bind (&ARDOUR_UI::update_sample_rate, this, _1), gui_context());
AudioEngine::instance()->Halted.connect_same_thread (halt_connection, boost::bind (&ARDOUR_UI::engine_halted, this, _1, false));
_tooltips.enable();
uint32_t samples = atoi (bs_text); /* will ignore trailing text */
uint32_t rate = get_rate();
- /* Translators: "msecs" is ALWAYS plural here, so we do not
- need singular form as well.
- */
/* Developers: note the hard-coding of a double buffered model
in the (2 * samples) computation of latency. we always start
the audiobackend in this configuration.
*/
+ /* note to jack1 developers: ardour also always starts the engine
+ * in async mode (no jack2 --sync option) which adds an extra cycle
+ * of latency with jack2 (and *3 would be correct)
+ * The value can also be wrong if jackd is started externally..
+ *
+ * At the time of writing the ALSA backend always uses double-buffering *2,
+ * The Dummy backend *1, and who knows what ASIO really does :)
+ *
+ * So just display the period size, that's also what
+ * ARDOUR_UI::update_sample_rate() does for the status bar.
+ * (the statusbar calls AudioEngine::instance()->usecs_per_cycle()
+ * but still, that's the buffer period, not [round-trip] latency)
+ */
char buf[32];
- snprintf (buf, sizeof (buf), _("(%.1f msecs)"), (2 * samples) / (rate/1000.0));
+ snprintf (buf, sizeof (buf), _("(%.1f ms)"), (samples / (rate/1000.0f)));
buffer_size_duration_label.set_text (buf);
}