--- a/gst_plugins_base/gst-libs/gst/audio/gstbaseaudiosrc.c Tue Aug 31 15:30:33 2010 +0300
+++ b/gst_plugins_base/gst-libs/gst/audio/gstbaseaudiosrc.c Wed Sep 01 12:16:41 2010 +0100
@@ -42,31 +42,12 @@
#include "gst/gst-i18n-plugin.h"
-GST_DEBUG_CATEGORY_STATIC (gst_base_audio_src_debug);
-#define GST_CAT_DEFAULT gst_base_audio_src_debug
-
#ifdef __SYMBIAN32__
-EXPORT_C
+#include <glib_global.h>
#endif
-GType
-gst_base_audio_src_slave_method_get_type (void)
-{
- static GType slave_method_type = 0;
- static const GEnumValue slave_method[] = {
- {GST_BASE_AUDIO_SRC_SLAVE_RESAMPLE, "Resampling slaving", "resample"},
- {GST_BASE_AUDIO_SRC_SLAVE_RETIMESTAMP, "Re-timestamp", "re-timestamp"},
- {GST_BASE_AUDIO_SRC_SLAVE_SKEW, "Skew", "skew"},
- {GST_BASE_AUDIO_SRC_SLAVE_NONE, "No slaving", "none"},
- {0, NULL, NULL},
- };
-
- if (!slave_method_type) {
- slave_method_type =
- g_enum_register_static ("GstBaseAudioSrcSlaveMethod", slave_method);
- }
- return slave_method_type;
-}
+GST_DEBUG_CATEGORY_STATIC (gst_base_audio_src_debug);
+#define GST_CAT_DEFAULT gst_base_audio_src_debug
#define GST_BASE_AUDIO_SRC_GET_PRIVATE(obj) \
(G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_BASE_AUDIO_SRC, GstBaseAudioSrcPrivate))
@@ -74,9 +55,6 @@
struct _GstBaseAudioSrcPrivate
{
gboolean provide_clock;
-
- /* the clock slaving algorithm in use */
- GstBaseAudioSrcSlaveMethod slave_method;
};
/* BaseAudioSrc signals and args */
@@ -88,21 +66,14 @@
#define DEFAULT_BUFFER_TIME ((200 * GST_MSECOND) / GST_USECOND)
#define DEFAULT_LATENCY_TIME ((10 * GST_MSECOND) / GST_USECOND)
-#define DEFAULT_ACTUAL_BUFFER_TIME -1
-#define DEFAULT_ACTUAL_LATENCY_TIME -1
#define DEFAULT_PROVIDE_CLOCK TRUE
-#define DEFAULT_SLAVE_METHOD GST_BASE_AUDIO_SRC_SLAVE_RETIMESTAMP
enum
{
PROP_0,
PROP_BUFFER_TIME,
PROP_LATENCY_TIME,
- PROP_ACTUAL_BUFFER_TIME,
- PROP_ACTUAL_LATENCY_TIME,
- PROP_PROVIDE_CLOCK,
- PROP_SLAVE_METHOD,
- PROP_LAST
+ PROP_PROVIDE_CLOCK
};
static void
@@ -115,7 +86,6 @@
GST_DEBUG ("binding text domain %s to locale dir %s", GETTEXT_PACKAGE,
LOCALEDIR);
bindtextdomain (GETTEXT_PACKAGE, LOCALEDIR);
- bind_textdomain_codeset (GETTEXT_PACKAGE, "UTF-8");
#endif /* ENABLE_NLS */
}
@@ -177,51 +147,17 @@
g_object_class_install_property (gobject_class, PROP_BUFFER_TIME,
g_param_spec_int64 ("buffer-time", "Buffer Time",
"Size of audio buffer in microseconds", 1,
- G_MAXINT64, DEFAULT_BUFFER_TIME,
- G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+ G_MAXINT64, DEFAULT_BUFFER_TIME, G_PARAM_READWRITE));
g_object_class_install_property (gobject_class, PROP_LATENCY_TIME,
g_param_spec_int64 ("latency-time", "Latency Time",
"Audio latency in microseconds", 1,
- G_MAXINT64, DEFAULT_LATENCY_TIME,
- G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
-
- /**
- * GstBaseAudioSrc:actual-buffer-time:
- *
- * Actual configured size of audio buffer in microseconds.
- *
- * Since: 0.10.20
- **/
- g_object_class_install_property (gobject_class, PROP_ACTUAL_BUFFER_TIME,
- g_param_spec_int64 ("actual-buffer-time", "Actual Buffer Time",
- "Actual configured size of audio buffer in microseconds",
- DEFAULT_ACTUAL_BUFFER_TIME, G_MAXINT64, DEFAULT_ACTUAL_BUFFER_TIME,
- G_PARAM_READABLE));
-
- /**
- * GstBaseAudioSrc:actual-latency-time:
- *
- * Actual configured audio latency in microseconds.
- *
- * Since: 0.10.20
- **/
- g_object_class_install_property (gobject_class, PROP_ACTUAL_LATENCY_TIME,
- g_param_spec_int64 ("actual-latency-time", "Actual Latency Time",
- "Actual configured audio latency in microseconds",
- DEFAULT_ACTUAL_LATENCY_TIME, G_MAXINT64, DEFAULT_ACTUAL_LATENCY_TIME,
- G_PARAM_READABLE));
+ G_MAXINT64, DEFAULT_LATENCY_TIME, G_PARAM_READWRITE));
g_object_class_install_property (gobject_class, PROP_PROVIDE_CLOCK,
g_param_spec_boolean ("provide-clock", "Provide Clock",
"Provide a clock to be used as the global pipeline clock",
- DEFAULT_PROVIDE_CLOCK, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
-
- g_object_class_install_property (gobject_class, PROP_SLAVE_METHOD,
- g_param_spec_enum ("slave-method", "Slave Method",
- "Algorithm to use to match the rate of the masterclock",
- GST_TYPE_BASE_AUDIO_SRC_SLAVE_METHOD, DEFAULT_SLAVE_METHOD,
- G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+ DEFAULT_PROVIDE_CLOCK, G_PARAM_READWRITE));
gstelement_class->change_state =
GST_DEBUG_FUNCPTR (gst_base_audio_src_change_state);
@@ -241,7 +177,6 @@
/* ref class from a thread-safe context to work around missing bit of
* thread-safety in GObject */
g_type_class_ref (GST_TYPE_AUDIO_CLOCK);
- g_type_class_ref (GST_TYPE_RING_BUFFER);
}
static void
@@ -253,7 +188,6 @@
baseaudiosrc->buffer_time = DEFAULT_BUFFER_TIME;
baseaudiosrc->latency_time = DEFAULT_LATENCY_TIME;
baseaudiosrc->priv->provide_clock = DEFAULT_PROVIDE_CLOCK;
- baseaudiosrc->priv->slave_method = DEFAULT_SLAVE_METHOD;
/* reset blocksize we use latency time to calculate a more useful
* value based on negotiated format. */
GST_BASE_SRC (baseaudiosrc)->blocksize = 0;
@@ -274,7 +208,6 @@
src = GST_BASE_AUDIO_SRC (object);
- GST_OBJECT_LOCK (src);
if (src->clock)
gst_object_unref (src->clock);
src->clock = NULL;
@@ -283,7 +216,6 @@
gst_object_unparent (GST_OBJECT_CAST (src->ringbuffer));
src->ringbuffer = NULL;
}
- GST_OBJECT_UNLOCK (src);
G_OBJECT_CLASS (parent_class)->dispose (object);
}
@@ -418,58 +350,6 @@
return result;
}
-/**
- * gst_base_audio_src_set_slave_method:
- * @src: a #GstBaseAudioSrc
- * @method: the new slave method
- *
- * Controls how clock slaving will be performed in @src.
- *
- * Since: 0.10.20
- */
-#ifdef __SYMBIAN32__
-EXPORT_C
-#endif
-
-void
-gst_base_audio_src_set_slave_method (GstBaseAudioSrc * src,
- GstBaseAudioSrcSlaveMethod method)
-{
- g_return_if_fail (GST_IS_BASE_AUDIO_SRC (src));
-
- GST_OBJECT_LOCK (src);
- src->priv->slave_method = method;
- GST_OBJECT_UNLOCK (src);
-}
-
-/**
- * gst_base_audio_src_get_slave_method:
- * @src: a #GstBaseAudioSrc
- *
- * Get the current slave method used by @src.
- *
- * Returns: The current slave method used by @src.
- *
- * Since: 0.10.20
- */
-#ifdef __SYMBIAN32__
-EXPORT_C
-#endif
-
-GstBaseAudioSrcSlaveMethod
-gst_base_audio_src_get_slave_method (GstBaseAudioSrc * src)
-{
- GstBaseAudioSrcSlaveMethod result;
-
- g_return_val_if_fail (GST_IS_BASE_AUDIO_SRC (src), -1);
-
- GST_OBJECT_LOCK (src);
- result = src->priv->slave_method;
- GST_OBJECT_UNLOCK (src);
-
- return result;
-}
-
static void
gst_base_audio_src_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
@@ -488,9 +368,6 @@
case PROP_PROVIDE_CLOCK:
gst_base_audio_src_set_provide_clock (src, g_value_get_boolean (value));
break;
- case PROP_SLAVE_METHOD:
- gst_base_audio_src_set_slave_method (src, g_value_get_enum (value));
- break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
@@ -512,28 +389,9 @@
case PROP_LATENCY_TIME:
g_value_set_int64 (value, src->latency_time);
break;
- case PROP_ACTUAL_BUFFER_TIME:
- GST_OBJECT_LOCK (src);
- if (src->ringbuffer && src->ringbuffer->acquired)
- g_value_set_int64 (value, src->ringbuffer->spec.buffer_time);
- else
- g_value_set_int64 (value, DEFAULT_ACTUAL_BUFFER_TIME);
- GST_OBJECT_UNLOCK (src);
- break;
- case PROP_ACTUAL_LATENCY_TIME:
- GST_OBJECT_LOCK (src);
- if (src->ringbuffer && src->ringbuffer->acquired)
- g_value_set_int64 (value, src->ringbuffer->spec.latency_time);
- else
- g_value_set_int64 (value, DEFAULT_ACTUAL_LATENCY_TIME);
- GST_OBJECT_UNLOCK (src);
- break;
case PROP_PROVIDE_CLOCK:
g_value_set_boolean (value, gst_base_audio_src_get_provide_clock (src));
break;
- case PROP_SLAVE_METHOD:
- g_value_set_enum (value, gst_base_audio_src_get_slave_method (src));
- break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
@@ -605,9 +463,6 @@
gst_ring_buffer_debug_spec_buff (spec);
- g_object_notify (G_OBJECT (src), "actual-buffer-time");
- g_object_notify (G_OBJECT (src), "actual-latency-time");
-
return TRUE;
/* ERRORS */
@@ -683,31 +538,21 @@
gst_base_audio_src_event (GstBaseSrc * bsrc, GstEvent * event)
{
GstBaseAudioSrc *src = GST_BASE_AUDIO_SRC (bsrc);
- gboolean res;
-
- res = TRUE;
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_FLUSH_START:
- GST_DEBUG_OBJECT (bsrc, "flush-start");
gst_ring_buffer_pause (src->ringbuffer);
gst_ring_buffer_clear_all (src->ringbuffer);
break;
case GST_EVENT_FLUSH_STOP:
- GST_DEBUG_OBJECT (bsrc, "flush-stop");
/* always resync on sample after a flush */
src->next_sample = -1;
gst_ring_buffer_clear_all (src->ringbuffer);
break;
- case GST_EVENT_SEEK:
- GST_DEBUG_OBJECT (bsrc, "refuse to seek");
- res = FALSE;
- break;
default:
- GST_DEBUG_OBJECT (bsrc, "dropping event %p", event);
break;
}
- return res;
+ return TRUE;
}
/* get the next offset in the ringbuffer for reading samples.
@@ -746,7 +591,7 @@
if (diff >= segtotal) {
GST_DEBUG_OBJECT (src, "dropped, align to segment %d", segdone);
/* sample would be dropped, position to next playable position */
- sample = ((guint64) (segdone)) * sps;
+ sample = (segdone - segtotal + 1) * sps;
}
return sample;
@@ -832,9 +677,7 @@
G_GUINT64_FORMAT, sample - src->next_sample, sample);
GST_ELEMENT_WARNING (src, CORE, CLOCK,
(_("Can't record audio fast enough")),
- ("Dropped %" G_GUINT64_FORMAT " samples. This is most likely because "
- "downstream can't keep up and is consuming samples too slowly.",
- sample - src->next_sample));
+ ("dropped %" G_GUINT64_FORMAT " samples", sample - src->next_sample));
GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_DISCONT);
}
@@ -846,179 +689,28 @@
spec->rate) - timestamp;
GST_OBJECT_LOCK (src);
- if (!(clock = GST_ELEMENT_CLOCK (src)))
- goto no_sync;
-
- if (clock != src->clock) {
- /* we are slaved, check how to handle this */
- switch (src->priv->slave_method) {
- case GST_BASE_AUDIO_SRC_SLAVE_RESAMPLE:
- /* not implemented, use skew algorithm. This algorithm should
- * work on the readout pointer and produces more or less samples based
- * on the clock drift */
- case GST_BASE_AUDIO_SRC_SLAVE_SKEW:
- {
- GstClockTime running_time;
- GstClockTime base_time;
- GstClockTime current_time;
- guint64 running_time_sample;
- gint running_time_segment;
- gint current_segment;
- gint segment_skew;
- gint sps;
-
- /* samples per segment */
- sps = ringbuffer->samples_per_seg;
-
- /* get the current time */
- current_time = gst_clock_get_time (clock);
-
- /* get the basetime */
- base_time = GST_ELEMENT_CAST (src)->base_time;
-
- /* get the running_time */
- running_time = current_time - base_time;
-
- /* the running_time converted to a sample (relative to the ringbuffer) */
- running_time_sample =
- gst_util_uint64_scale_int (running_time, spec->rate, GST_SECOND);
-
- /* the segmentnr corrensponding to running_time, round down */
- running_time_segment = running_time_sample / sps;
-
- /* the segment currently read from the ringbuffer */
- current_segment = sample / sps;
-
- /* the skew we have between running_time and the ringbuffertime */
- segment_skew = running_time_segment - current_segment;
-
- GST_DEBUG_OBJECT (bsrc, "\n running_time = %" GST_TIME_FORMAT
- "\n timestamp = %" GST_TIME_FORMAT
- "\n running_time_segment = %d"
- "\n current_segment = %d"
- "\n segment_skew = %d",
- GST_TIME_ARGS (running_time),
- GST_TIME_ARGS (timestamp),
- running_time_segment, current_segment, segment_skew);
+ clock = GST_ELEMENT_CLOCK (src);
+ if (clock != NULL && clock != src->clock) {
+ GstClockTime base_time, latency;
- /* Resync the ringbuffer if:
- * 1. We get one segment into the future.
- * This is clearly a lie, because we can't
- * possibly have a buffer with timestamp 1 at
- * time 0. (unless it has time-travelled...)
- *
- * 2. We are more than the length of the ringbuffer behind.
- * The length of the ringbuffer then gets to dictate
- * the threshold for what is concidered "too late"
- *
- * 3. If this is our first buffer.
- * We know that we should catch up to running_time
- * the first time we are ran.
- */
- if ((segment_skew < 0) ||
- (segment_skew >= ringbuffer->spec.segtotal) ||
- (current_segment == 0)) {
- gint segments_written;
- gint first_segment;
- gint last_segment;
- gint new_last_segment;
- gint segment_diff;
- gint new_first_segment;
- guint64 new_sample;
-
- /* we are going to say that the last segment was captured at the current time
- (running_time), minus one segment of creation-latency in the ringbuffer.
- This can be thought of as: The segment arrived in the ringbuffer at time X, and
- that means it was created at time X - (one segment). */
- new_last_segment = running_time_segment - 1;
-
- /* for better readablity */
- first_segment = current_segment;
-
- /* get the amount of segments written from the device by now */
- segments_written = g_atomic_int_get (&ringbuffer->segdone);
-
- /* subtract the base to segments_written to get the number of the
- last written segment in the ringbuffer (one segment written = segment 0) */
- last_segment = segments_written - ringbuffer->segbase - 1;
-
- /* we see how many segments the ringbuffer was timeshifted */
- segment_diff = new_last_segment - last_segment;
-
- /* we move the first segment an equal amount */
- new_first_segment = first_segment + segment_diff;
-
- /* and we also move the segmentbase the same amount */
- ringbuffer->segbase -= segment_diff;
-
- /* we calculate the new sample value */
- new_sample = ((guint64) new_first_segment) * sps;
-
- /* and get the relative time to this -> our new timestamp */
- timestamp =
- gst_util_uint64_scale_int (new_sample, GST_SECOND, spec->rate);
-
- /* we update the next sample accordingly */
- src->next_sample = new_sample + samples;
-
- GST_DEBUG_OBJECT (bsrc,
- "Timeshifted the ringbuffer with %d segments: "
- "Updating the timestamp to %" GST_TIME_FORMAT ", "
- "and src->next_sample to %" G_GUINT64_FORMAT, segment_diff,
- GST_TIME_ARGS (timestamp), src->next_sample);
- }
- break;
- }
- case GST_BASE_AUDIO_SRC_SLAVE_RETIMESTAMP:
- {
- GstClockTime base_time, latency;
-
- /* We are slaved to another clock, take running time of the pipeline clock and
- * timestamp against it. Somebody else in the pipeline should figure out the
- * clock drift. We keep the duration we calculated above. */
- timestamp = gst_clock_get_time (clock);
- base_time = GST_ELEMENT_CAST (src)->base_time;
-
- if (timestamp > base_time)
- timestamp -= base_time;
- else
- timestamp = 0;
-
- /* subtract latency */
- latency =
- gst_util_uint64_scale_int (total_samples, GST_SECOND, spec->rate);
- if (timestamp > latency)
- timestamp -= latency;
- else
- timestamp = 0;
- }
- case GST_BASE_AUDIO_SRC_SLAVE_NONE:
- break;
- }
- } else {
- GstClockTime base_time;
-
- /* to get the timestamp against the clock we also need to add our offset */
- timestamp = gst_audio_clock_adjust (clock, timestamp);
-
- /* we are not slaved, subtract base_time */
+ /* We are slaved to another clock, take running time of the clock and just
+ * timestamp against it. Somebody else in the pipeline should figure out the
+ * clock drift, for now. We keep the duration we calculated above. */
+ timestamp = gst_clock_get_time (clock);
base_time = GST_ELEMENT_CAST (src)->base_time;
- if (timestamp > base_time) {
+ if (timestamp > base_time)
timestamp -= base_time;
- GST_LOG_OBJECT (src,
- "buffer timestamp %" GST_TIME_FORMAT " (base_time %" GST_TIME_FORMAT
- ")", GST_TIME_ARGS (timestamp), GST_TIME_ARGS (base_time));
- } else {
- GST_LOG_OBJECT (src,
- "buffer timestamp 0, ts %" GST_TIME_FORMAT " <= base_time %"
- GST_TIME_FORMAT, GST_TIME_ARGS (timestamp),
- GST_TIME_ARGS (base_time));
+ else
timestamp = 0;
- }
+
+ /* subtract latency */
+ latency = gst_util_uint64_scale_int (total_samples, GST_SECOND, spec->rate);
+ if (timestamp > latency)
+ timestamp -= latency;
+ else
+ timestamp = 0;
}
-
-no_sync:
GST_OBJECT_UNLOCK (src);
GST_BUFFER_TIMESTAMP (buf) = timestamp;
@@ -1026,6 +718,8 @@
GST_BUFFER_OFFSET (buf) = sample;
GST_BUFFER_OFFSET_END (buf) = sample + samples;
+ gst_buffer_set_caps (buf, GST_PAD_CAPS (GST_BASE_SRC_PAD (bsrc)));
+
*outbuf = buf;
return GST_FLOW_OK;
@@ -1091,12 +785,9 @@
switch (transition) {
case GST_STATE_CHANGE_NULL_TO_READY:
GST_DEBUG_OBJECT (src, "NULL->READY");
- GST_OBJECT_LOCK (src);
if (src->ringbuffer == NULL) {
- gst_audio_clock_reset (GST_AUDIO_CLOCK (src->clock), 0);
src->ringbuffer = gst_base_audio_src_create_ringbuffer (src);
}
- GST_OBJECT_UNLOCK (src);
if (!gst_ring_buffer_open_device (src->ringbuffer))
goto open_failed;
break;
@@ -1133,10 +824,8 @@
case GST_STATE_CHANGE_READY_TO_NULL:
GST_DEBUG_OBJECT (src, "READY->NULL");
gst_ring_buffer_close_device (src->ringbuffer);
- GST_OBJECT_LOCK (src);
gst_object_unparent (GST_OBJECT_CAST (src->ringbuffer));
src->ringbuffer = NULL;
- GST_OBJECT_UNLOCK (src);
break;
default:
break;