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gst-libs/gst/audio/gstbaseaudiosink.c: Answer latency query.

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Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_class_init), (gst_base_audio_sink_query),
(gst_base_audio_sink_render), (gst_base_audio_sink_callback),
(gst_base_audio_sink_async_play),
(gst_base_audio_sink_change_state):
Answer latency query.
Use configured latency when syncing.
Fix clock slaving.
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_class_init), (gst_base_audio_src_dispose),
(gst_base_audio_src_query), (gst_base_audio_src_change_state):
Fix possible memleak.
Implement latency query.
Small cleanups.
This commit is contained in:
Wim Taymans 2007-02-15 12:06:25 +00:00
parent 70e52caf04
commit a43d0f57eb
3 changed files with 214 additions and 25 deletions
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18
ChangeLog
View file

@ -1,3 +1,21 @@
2007-02-15 Wim Taymans <wim@fluendo.com>
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_class_init), (gst_base_audio_sink_query),
(gst_base_audio_sink_render), (gst_base_audio_sink_callback),
(gst_base_audio_sink_async_play),
(gst_base_audio_sink_change_state):
Answer latency query.
Use configured latency when syncing.
Fix clock slaving.
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_class_init), (gst_base_audio_src_dispose),
(gst_base_audio_src_query), (gst_base_audio_src_change_state):
Fix possible memleak.
Implement latency query.
Small cleanups.
2007-02-15 Wim Taymans <wim@fluendo.com>
* ext/alsa/gstalsasink.c: (gst_alsasink_reset):

136
gst-libs/gst/audio/gstbaseaudiosink.c
View file

@ -87,6 +87,8 @@ static GstStateChangeReturn gst_base_audio_sink_change_state (GstElement *
element, GstStateChange transition);
static gboolean gst_base_audio_sink_activate_pull (GstBaseSink * basesink,
gboolean active);
static gboolean gst_base_audio_sink_query (GstElement * element, GstQuery *
query);
static GstClock *gst_base_audio_sink_provide_clock (GstElement * elem);
static GstClockTime gst_base_audio_sink_get_time (GstClock * clock,
@ -149,6 +151,7 @@ gst_base_audio_sink_class_init (GstBaseAudioSinkClass * klass)
GST_DEBUG_FUNCPTR (gst_base_audio_sink_change_state);
gstelement_class->provide_clock =
GST_DEBUG_FUNCPTR (gst_base_audio_sink_provide_clock);
gstelement_class->query = GST_DEBUG_FUNCPTR (gst_base_audio_sink_query);
gstbasesink_class->event = GST_DEBUG_FUNCPTR (gst_base_audio_sink_event);
gstbasesink_class->preroll = GST_DEBUG_FUNCPTR (gst_base_audio_sink_preroll);
@ -235,6 +238,62 @@ clock_disabled:
}
}
static gboolean
gst_base_audio_sink_query (GstElement * element, GstQuery * query)
{
gboolean res = FALSE;
GstBaseAudioSink *basesink = GST_BASE_AUDIO_SINK (element);
switch (GST_QUERY_TYPE (query)) {
case GST_QUERY_LATENCY:
{
gboolean live, us_live;
GstClockTime min_l, max_l;
GST_DEBUG_OBJECT (basesink, "latency query");
/* ask parent first, it will do an upstream query for us. */
if ((res =
gst_base_sink_query_latency (GST_BASE_SINK_CAST (basesink), &live,
&us_live, &min_l, &max_l))) {
GstClockTime min_latency, max_latency;
/* we and upstream are both live, adjust the min_latency */
if (live && us_live && basesink->ringbuffer
&& basesink->ringbuffer->spec.rate) {
GstRingBufferSpec *spec;
spec = &basesink->ringbuffer->spec;
max_latency =
spec->segtotal * spec->segsize * GST_SECOND / (spec->rate *
spec->bytes_per_sample);
min_latency = MAX (max_latency, min_l);
GST_DEBUG_OBJECT (basesink,
"peer min %" GST_TIME_FORMAT ", our min latency: %"
GST_TIME_FORMAT, GST_TIME_ARGS (min_l),
GST_TIME_ARGS (min_latency));
} else {
GST_DEBUG_OBJECT (basesink,
"peer or we are not live, don't care about latency");
min_latency = 0;
max_latency = -1;
}
gst_query_set_latency (query, live, min_latency, max_latency);
}
break;
}
default:
res = GST_ELEMENT_CLASS (parent_class)->query (element, query);
break;
}
return res;
}
static GstClockTime
gst_base_audio_sink_get_time (GstClock * clock, GstBaseAudioSink * sink)
{
@ -550,7 +609,7 @@ gst_base_audio_sink_get_offset (GstBaseAudioSink * sink)
static GstFlowReturn
gst_base_audio_sink_render (GstBaseSink * bsink, GstBuffer * buf)
{
guint64 in_offset, clock_offset;
guint64 in_offset;
GstClockTime time, stop, render_start, render_stop, sample_offset;
GstBaseAudioSink *sink;
GstRingBuffer *ringbuf;
@ -563,9 +622,9 @@ gst_base_audio_sink_render (GstBaseSink * bsink, GstBuffer * buf)
GstClockTime crate_num;
GstClockTime crate_denom;
gint out_samples;
GstClockTime cinternal, cexternal;
GstClockTime base_time, cinternal, cexternal, latency;
GstClock *clock;
gboolean sync;
gboolean sync, slaved;
sink = GST_BASE_AUDIO_SINK (bsink);
@ -663,29 +722,62 @@ gst_base_audio_sink_render (GstBaseSink * bsink, GstBuffer * buf)
"running: start %" GST_TIME_FORMAT " - stop %" GST_TIME_FORMAT,
GST_TIME_ARGS (render_start), GST_TIME_ARGS (render_stop));
/* get calibration parameters to compensate for speed and offset differences
* when we are slaved */
gst_clock_get_calibration (sink->provided_clock, &cinternal, &cexternal,
&crate_num, &crate_denom);
base_time = gst_element_get_base_time (GST_ELEMENT_CAST (bsink));
clock_offset =
(gst_element_get_base_time (GST_ELEMENT_CAST (bsink)) - cexternal) +
cinternal;
GST_DEBUG_OBJECT (sink, "base_time %" GST_TIME_FORMAT,
GST_TIME_ARGS (base_time));
GST_DEBUG_OBJECT (sink, "clock offset %" GST_TIME_FORMAT " %" G_GUINT64_FORMAT
"/%" G_GUINT64_FORMAT, GST_TIME_ARGS (clock_offset), crate_num,
crate_denom);
/* add base time to sync against the clock */
render_start += base_time;
render_stop += base_time;
/* and bring the time to the rate corrected offset in the buffer */
render_start = gst_util_uint64_scale_int (render_start + clock_offset,
ringbuf->spec.rate, GST_SECOND);
render_stop = gst_util_uint64_scale_int (render_stop + clock_offset,
ringbuf->spec.rate, GST_SECOND);
slaved = clock != sink->provided_clock;
if (slaved) {
/* get calibration parameters to compensate for speed and offset differences
* when we are slaved */
gst_clock_get_calibration (sink->provided_clock, &cinternal, &cexternal,
&crate_num, &crate_denom);
GST_DEBUG_OBJECT (sink, "internal %" GST_TIME_FORMAT " external %"
GST_TIME_FORMAT " %" G_GUINT64_FORMAT "/%" G_GUINT64_FORMAT " = %f",
GST_TIME_ARGS (cinternal), GST_TIME_ARGS (cexternal), crate_num,
crate_denom, (gdouble) crate_num / crate_denom);
/* bring to our slaved clock time */
if (render_start >= cexternal)
render_start =
gst_util_uint64_scale (render_start - cexternal, crate_denom,
crate_num) + cinternal;
else
render_start =
cinternal - gst_util_uint64_scale (cexternal - render_start,
crate_denom, crate_num);
if (render_stop >= cexternal)
render_stop =
gst_util_uint64_scale (render_stop - cexternal, crate_denom,
crate_num) + cinternal;
else
render_stop =
cinternal - gst_util_uint64_scale (cexternal - render_stop,
crate_denom, crate_num);
}
/* compensate for latency */
latency = gst_base_sink_get_latency (bsink);
render_start += latency;
render_stop += latency;
GST_DEBUG_OBJECT (sink,
"render: start %" GST_TIME_FORMAT " - stop %" GST_TIME_FORMAT,
GST_TIME_ARGS (render_start), GST_TIME_ARGS (render_stop));
/* and bring the time to the rate corrected offset in the buffer */
render_start = gst_util_uint64_scale_int (render_start,
ringbuf->spec.rate, GST_SECOND);
render_stop = gst_util_uint64_scale_int (render_stop,
ringbuf->spec.rate, GST_SECOND);
/* always resync after a discont */
if (G_UNLIKELY (GST_BUFFER_FLAG_IS_SET (buf, GST_BUFFER_FLAG_DISCONT))) {
GST_DEBUG_OBJECT (sink, "resync after discont");
@ -736,7 +828,7 @@ gst_base_audio_sink_render (GstBaseSink * bsink, GstBuffer * buf)
render_start += align;
/* only align stop if we are not slaved */
if (clock != sink->provided_clock) {
if (slaved) {
GST_DEBUG_OBJECT (sink, "no stop time align needed: we are slaved");
goto no_align;
}
@ -878,7 +970,8 @@ gst_base_audio_sink_callback (GstRingBuffer * rbuf, guint8 * data, guint len,
/* would be nice to arrange for pad_alloc_buffer to return data -- as it is we
will copy twice, once into data, once into DMA */
GST_LOG_OBJECT (basesink, "pulling %d bytes to fill audio buffer", len);
GST_LOG_OBJECT (basesink, "pulling %d bytes offset %" G_GUINT64_FORMAT
" to fill audio buffer", len, basesink->offset);
ret = gst_pad_pull_range (basesink->sinkpad, basesink->offset, len, &buf);
if (ret != GST_FLOW_OK)
goto error;
@ -944,6 +1037,8 @@ gst_base_audio_sink_async_play (GstBaseSink * basesink)
}
no_clock:
gst_ring_buffer_start (sink->ringbuffer);
return GST_STATE_CHANGE_SUCCESS;
}
@ -1009,7 +1104,6 @@ gst_base_audio_sink_change_state (GstElement * element,
break;
case GST_STATE_CHANGE_PAUSED_TO_READY:
gst_ring_buffer_release (sink->ringbuffer);
gst_pad_set_caps (GST_BASE_SINK_PAD (sink), NULL);
break;
case GST_STATE_CHANGE_READY_TO_NULL:
gst_ring_buffer_close_device (sink->ringbuffer);

85
gst-libs/gst/audio/gstbaseaudiosrc.c
View file

@ -66,6 +66,7 @@ static void gst_base_audio_src_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void gst_base_audio_src_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static void gst_base_audio_src_dispose (GObject * object);
static GstStateChangeReturn gst_base_audio_src_change_state (GstElement *
element, GstStateChange transition);
@ -84,6 +85,7 @@ static gboolean gst_base_audio_src_event (GstBaseSrc * bsrc, GstEvent * event);
static void gst_base_audio_src_get_times (GstBaseSrc * bsrc,
GstBuffer * buffer, GstClockTime * start, GstClockTime * end);
static gboolean gst_base_audio_src_setcaps (GstBaseSrc * bsrc, GstCaps * caps);
static gboolean gst_base_audio_src_query (GstBaseSrc * bsrc, GstQuery * query);
static void gst_base_audio_src_fixate (GstBaseSrc * bsrc, GstCaps * caps);
/* static guint gst_base_audio_src_signals[LAST_SIGNAL] = { 0 }; */
@ -110,6 +112,7 @@ gst_base_audio_src_class_init (GstBaseAudioSrcClass * klass)
GST_DEBUG_FUNCPTR (gst_base_audio_src_set_property);
gobject_class->get_property =
GST_DEBUG_FUNCPTR (gst_base_audio_src_get_property);
gobject_class->dispose = GST_DEBUG_FUNCPTR (gst_base_audio_src_dispose);
g_object_class_install_property (gobject_class, PROP_BUFFER_TIME,
g_param_spec_int64 ("buffer-time", "Buffer Time",
@ -130,6 +133,7 @@ gst_base_audio_src_class_init (GstBaseAudioSrcClass * klass)
gstbasesrc_class->set_caps = GST_DEBUG_FUNCPTR (gst_base_audio_src_setcaps);
gstbasesrc_class->event = GST_DEBUG_FUNCPTR (gst_base_audio_src_event);
gstbasesrc_class->query = GST_DEBUG_FUNCPTR (gst_base_audio_src_query);
gstbasesrc_class->get_times =
GST_DEBUG_FUNCPTR (gst_base_audio_src_get_times);
gstbasesrc_class->create = GST_DEBUG_FUNCPTR (gst_base_audio_src_create);
@ -157,6 +161,25 @@ gst_base_audio_src_init (GstBaseAudioSrc * baseaudiosrc,
gst_base_src_set_format (GST_BASE_SRC (baseaudiosrc), GST_FORMAT_TIME);
}
static void
gst_base_audio_src_dispose (GObject * object)
{
GstBaseAudioSrc *src;
src = GST_BASE_AUDIO_SRC (object);
if (src->clock)
gst_object_unref (src->clock);
src->clock = NULL;
if (src->ringbuffer) {
gst_object_unparent (GST_OBJECT_CAST (src->ringbuffer));
src->ringbuffer = NULL;
}
G_OBJECT_CLASS (parent_class)->dispose (object);
}
static gboolean
gst_base_audio_src_set_clock (GstElement * elem, GstClock * clock)
{
@ -378,6 +401,50 @@ gst_base_audio_src_get_times (GstBaseSrc * bsrc, GstBuffer * buffer,
*end = GST_CLOCK_TIME_NONE;
}
static gboolean
gst_base_audio_src_query (GstBaseSrc * bsrc, GstQuery * query)
{
GstBaseAudioSrc *src = GST_BASE_AUDIO_SRC (bsrc);
gboolean res = FALSE;
switch (GST_QUERY_TYPE (query)) {
case GST_QUERY_LATENCY:
{
GstClockTime min_latency, max_latency;
GstRingBufferSpec *spec;
if (G_UNLIKELY (src->ringbuffer == NULL
|| src->ringbuffer->spec.rate == 0))
goto done;
spec = &src->ringbuffer->spec;
min_latency =
gst_util_uint64_scale_int (spec->segsize, GST_SECOND,
spec->rate * spec->bytes_per_sample);
max_latency =
gst_util_uint64_scale_int (spec->segtotal * spec->segsize, GST_SECOND,
spec->rate * spec->bytes_per_sample);
GST_DEBUG_OBJECT (src,
"report latency min %" GST_TIME_FORMAT " max %" GST_TIME_FORMAT,
GST_TIME_ARGS (min_latency), GST_TIME_ARGS (max_latency));
/* we are always live, the min latency is 1 segment and the max latency is
* the complete buffer of segments. */
gst_query_set_latency (query, TRUE, min_latency, max_latency);
res = TRUE;
break;
}
default:
res = GST_BASE_SRC_CLASS (parent_class)->query (bsrc, query);
break;
}
done:
return res;
}
static gboolean
gst_base_audio_src_event (GstBaseSrc * bsrc, GstEvent * event)
{
@ -588,15 +655,18 @@ gst_base_audio_src_change_state (GstElement * element,
src->ringbuffer = gst_base_audio_src_create_ringbuffer (src);
}
if (!gst_ring_buffer_open_device (src->ringbuffer))
return GST_STATE_CHANGE_FAILURE;
src->next_sample = -1;
goto open_failed;
break;
case GST_STATE_CHANGE_READY_TO_PAUSED:
src->next_sample = -1;
gst_ring_buffer_set_flushing (src->ringbuffer, FALSE);
break;
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
gst_ring_buffer_may_start (src->ringbuffer, TRUE);
break;
case GST_STATE_CHANGE_PAUSED_TO_READY:
gst_ring_buffer_set_flushing (src->ringbuffer, TRUE);
break;
default:
break;
}
@ -609,9 +679,7 @@ gst_base_audio_src_change_state (GstElement * element,
gst_ring_buffer_pause (src->ringbuffer);
break;
case GST_STATE_CHANGE_PAUSED_TO_READY:
gst_ring_buffer_set_flushing (src->ringbuffer, TRUE);
gst_ring_buffer_release (src->ringbuffer);
src->next_sample = -1;
break;
case GST_STATE_CHANGE_READY_TO_NULL:
gst_ring_buffer_close_device (src->ringbuffer);
@ -623,4 +691,13 @@ gst_base_audio_src_change_state (GstElement * element,
}
return ret;
/* ERRORS */
open_failed:
{
/* subclass must post a meaningfull error message */
GST_DEBUG_OBJECT (src, "open failed");
return GST_STATE_CHANGE_FAILURE;
}
}