Commit 0be7d56e authored by Wim Taymans's avatar Wim Taymans
Browse files

gst-libs/gst/audio/gstbaseaudiosink.c: Use scale functions when possible.

Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_get_time), (gst_base_audio_sink_setcaps),
(gst_base_audio_sink_drain), (gst_base_audio_sink_preroll),
(gst_base_audio_sink_render), (gst_base_audio_sink_change_state):
Use scale functions when possible.
Fix error messages.
Free clockid when after waiting for EOS.
Use G_(UN_)LIKLY when it makes sense.
Fix sample clipping bug found by Arwed v. Merkatz fixes #330789.
parent 50951316
2006-02-12 Wim Taymans <wim@fluendo.com>
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_get_time), (gst_base_audio_sink_setcaps),
(gst_base_audio_sink_drain), (gst_base_audio_sink_preroll),
(gst_base_audio_sink_render), (gst_base_audio_sink_change_state):
Use scale functions when possible.
Fix error messages.
Free clockid when after waiting for EOS.
Use G_(UN_)LIKLY when it makes sense.
Fix sample clipping bug found by Arwed v. Merkatz fixes #330789.
2006-02-12 Edward Hervey <edward@fluendo.com>
 
* gst/playback/gstplaybasebin.c: (prepare_output):
......
......@@ -218,7 +218,8 @@ gst_base_audio_sink_get_time (GstClock * clock, GstBaseAudioSink * sink)
/* our processed samples are always increasing */
samples = gst_ring_buffer_samples_done (sink->ringbuffer);
result = samples * GST_SECOND / sink->ringbuffer->spec.rate;
result = gst_util_uint64_scale_int (samples, GST_SECOND,
sink->ringbuffer->spec.rate);
return result;
}
......@@ -320,7 +321,7 @@ parse_error:
{
GST_DEBUG_OBJECT (sink, "could not parse caps");
GST_ELEMENT_ERROR (sink, STREAM, FORMAT,
("cannot parse audio format."), ("cannot parse audio format."));
(NULL), ("cannot parse audio format."));
return FALSE;
}
acquire_error:
......@@ -340,6 +341,9 @@ gst_base_audio_sink_get_times (GstBaseSink * bsink, GstBuffer * buffer,
*end = GST_CLOCK_TIME_NONE;
}
/* FIXME, this waits for the drain to happen but it cannot be
* canceled.
*/
static gboolean
gst_base_audio_sink_drain (GstBaseAudioSink * sink)
{
......@@ -364,6 +368,8 @@ gst_base_audio_sink_drain (GstBaseAudioSink * sink)
GST_DEBUG_OBJECT (sink, "waiting for last sample to play");
gst_clock_id_wait (id, NULL);
gst_clock_id_unref (id);
sink->next_sample = -1;
} else {
GST_OBJECT_UNLOCK (sink);
......@@ -414,8 +420,7 @@ gst_base_audio_sink_preroll (GstBaseSink * bsink, GstBuffer * buffer)
wrong_state:
{
GST_DEBUG_OBJECT (sink, "ringbuffer in wrong state");
GST_ELEMENT_ERROR (sink, RESOURCE, NOT_FOUND,
("sink not negotiated."), ("sink not negotiated."));
GST_ELEMENT_ERROR (sink, STREAM, FORMAT, (NULL), ("sink not negotiated."));
return GST_FLOW_NOT_NEGOTIATED;
}
}
......@@ -469,7 +474,7 @@ gst_base_audio_sink_render (GstBaseSink * bsink, GstBuffer * buf)
sink = GST_BASE_AUDIO_SINK (bsink);
if (GST_BUFFER_FLAG_IS_SET (buf, GST_BUFFER_FLAG_DISCONT)) {
if (G_UNLIKELY (GST_BUFFER_FLAG_IS_SET (buf, GST_BUFFER_FLAG_DISCONT))) {
/* always resync after a discont */
sink->next_sample = -1;
}
......@@ -477,13 +482,13 @@ gst_base_audio_sink_render (GstBaseSink * bsink, GstBuffer * buf)
ringbuf = sink->ringbuffer;
/* can't do anything when we don't have the device */
if (!gst_ring_buffer_is_acquired (ringbuf))
if (G_UNLIKELY (!gst_ring_buffer_is_acquired (ringbuf)))
goto wrong_state;
bps = ringbuf->spec.bytes_per_sample;
size = GST_BUFFER_SIZE (buf);
if (size % bps != 0)
if (G_UNLIKELY (size % bps) != 0)
goto wrong_size;
samples = size / bps;
......@@ -524,15 +529,17 @@ gst_base_audio_sink_render (GstBaseSink * bsink, GstBuffer * buf)
/* see if some clipping happened */
diff = ctime - time;
if (diff > 0) {
/* bring clipped time to samples */
diff = gst_util_uint64_scale_int (diff, ringbuf->spec.rate, GST_SECOND);
GST_DEBUG_OBJECT (sink, "clipping start to %" GST_TIME_FORMAT " %"
G_GUINT64_FORMAT " samples", GST_TIME_ARGS (ctime), diff);
samples -= diff;
data += samples * bps;
data += diff * bps;
time = ctime;
}
diff = stop - cstop;
if (diff > 0) {
/* bring clipped time to samples */
diff = gst_util_uint64_scale_int (diff, ringbuf->spec.rate, GST_SECOND);
GST_DEBUG_OBJECT (sink, "clipping stop to %" GST_TIME_FORMAT " %"
G_GUINT64_FORMAT " samples", GST_TIME_ARGS (cstop), diff);
......@@ -559,7 +566,7 @@ gst_base_audio_sink_render (GstBaseSink * bsink, GstBuffer * buf)
GST_TIME_ARGS (render_time), render_offset, samples);
/* roundoff errors in timestamp conversion */
if (sink->next_sample != -1) {
if (G_LIKELY (sink->next_sample != -1)) {
diff = ABS ((gint64) render_offset - (gint64) sink->next_sample);
/* we tollerate a 10th of a second diff before we start resyncing. This
......@@ -622,6 +629,7 @@ no_sync:
return GST_FLOW_OK;
/* SPECIAL cases */
out_of_segment:
{
GST_DEBUG_OBJECT (sink,
......@@ -630,19 +638,18 @@ out_of_segment:
GST_TIME_ARGS (bsink->segment.start));
return GST_FLOW_OK;
}
/* ERRORS */
wrong_state:
{
GST_DEBUG_OBJECT (sink, "ringbuffer not negotiated");
GST_ELEMENT_ERROR (sink, RESOURCE, NOT_FOUND,
("sink not negotiated."), ("sink not negotiated."));
GST_ELEMENT_ERROR (sink, STREAM, FORMAT, (NULL), ("sink not negotiated."));
return GST_FLOW_NOT_NEGOTIATED;
}
wrong_size:
{
GST_DEBUG_OBJECT (sink, "wrong size");
GST_ELEMENT_ERROR (sink, RESOURCE, NOT_FOUND,
("sink received buffer of wrong size."),
("sink received buffer of wrong size."));
GST_ELEMENT_ERROR (sink, STREAM, WRONG_TYPE,
(NULL), ("sink received buffer of wrong size."));
return GST_FLOW_ERROR;
}
stopping:
......@@ -690,7 +697,7 @@ gst_base_audio_sink_change_state (GstElement * element,
gst_base_audio_sink_callback, sink);
}
if (!gst_ring_buffer_open_device (sink->ringbuffer))
return GST_STATE_CHANGE_FAILURE;
goto open_failed;
sink->next_sample = 0;
break;
case GST_STATE_CHANGE_READY_TO_PAUSED:
......@@ -699,20 +706,25 @@ gst_base_audio_sink_change_state (GstElement * element,
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
{
GstClock *clock;
GstClockTime time, base;
/* FIXME, only start slaving when we really start the ringbuffer */
GST_OBJECT_LOCK (sink);
clock = GST_ELEMENT_CLOCK (sink);
if (clock == NULL)
goto no_clock;
/* FIXME, only start slaving when we really start the ringbuffer */
/* if we are slaved to a clock, we need to set the initial
* calibration */
if (clock != sink->provided_clock) {
GstClockTime time;
GstClockTime rate_num, rate_denom;
base = element->base_time;
time = gst_clock_get_internal_time (sink->provided_clock);
GST_DEBUG_OBJECT (sink, "time: %" GST_TIME_FORMAT,
GST_TIME_ARGS (time));
GST_DEBUG_OBJECT (sink,
"time: %" GST_TIME_FORMAT " base: %" GST_TIME_FORMAT,
GST_TIME_ARGS (time), GST_TIME_ARGS (base));
gst_clock_set_master (sink->provided_clock, clock);
/* FIXME, this is not yet accurate enough for smooth playback */
......@@ -722,6 +734,7 @@ gst_base_audio_sink_change_state (GstElement * element,
gst_clock_set_calibration (sink->provided_clock,
time, element->base_time, rate_num, rate_denom);
}
no_clock:
GST_OBJECT_UNLOCK (sink);
break;
}
......@@ -754,4 +767,10 @@ gst_base_audio_sink_change_state (GstElement * element,
}
return ret;
open_failed:
{
GST_DEBUG_OBJECT (sink, "open failed");
return GST_STATE_CHANGE_FAILURE;
}
}
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