LAST_SIGNAL
};
-#define DEFAULT_BUFFER_TIME 500 * GST_USECOND
-#define DEFAULT_LATENCY_TIME 10 * GST_USECOND
+#define DEFAULT_BUFFER_TIME ((200 * GST_MSECOND) / GST_USECOND)
+#define DEFAULT_LATENCY_TIME ((10 * GST_MSECOND) / GST_USECOND)
+
enum
{
PROP_0,
element, GstStateChange transition);
static GstClock *gst_base_audio_src_provide_clock (GstElement * elem);
+static gboolean gst_base_audio_src_set_clock (GstElement * elem,
+ GstClock * clock);
static GstClockTime gst_base_audio_src_get_time (GstClock * clock,
GstBaseAudioSrc * src);
-static GstFlowReturn gst_base_audio_src_create (GstPushSrc * psrc,
- GstBuffer ** buf);
+static GstFlowReturn gst_base_audio_src_create (GstBaseSrc * bsrc,
+ guint64 offset, guint length, GstBuffer ** buf);
+static gboolean gst_base_audio_src_check_get_range (GstBaseSrc * bsrc);
static gboolean gst_base_audio_src_event (GstBaseSrc * bsrc, GstEvent * event);
static void gst_base_audio_src_get_times (GstBaseSrc * bsrc,
static void
gst_base_audio_src_class_init (GstBaseAudioSrcClass * klass)
{
- gchar *longdesc;
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
GstBaseSrcClass *gstbasesrc_class;
gobject_class->get_property =
GST_DEBUG_FUNCPTR (gst_base_audio_src_get_property);
- longdesc =
- g_strdup_printf
- ("Size of audio buffer in microseconds (use -1 for default of %"
- G_GUINT64_FORMAT " us)", DEFAULT_BUFFER_TIME / GST_USECOND);
- g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_BUFFER_TIME,
- g_param_spec_int64 ("buffer-time", "Buffer Time", longdesc, -1,
+ 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_free (longdesc);
- longdesc =
- g_strdup_printf ("Audio latency in microseconds (use -1 for default of %"
- G_GUINT64_FORMAT " us)", DEFAULT_LATENCY_TIME / GST_USECOND);
- g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_LATENCY_TIME,
- g_param_spec_int64 ("latency-time", "Latency Time", longdesc, -1,
+
+ 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_free (longdesc);
gstelement_class->change_state =
GST_DEBUG_FUNCPTR (gst_base_audio_src_change_state);
gstelement_class->provide_clock =
GST_DEBUG_FUNCPTR (gst_base_audio_src_provide_clock);
+ gstelement_class->set_clock =
+ GST_DEBUG_FUNCPTR (gst_base_audio_src_set_clock);
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->get_times =
GST_DEBUG_FUNCPTR (gst_base_audio_src_get_times);
-
- gstpushsrc_class->create = GST_DEBUG_FUNCPTR (gst_base_audio_src_create);
+ gstbasesrc_class->create = GST_DEBUG_FUNCPTR (gst_base_audio_src_create);
+ gstbasesrc_class->check_get_range =
+ GST_DEBUG_FUNCPTR (gst_base_audio_src_check_get_range);
}
static void
{
baseaudiosrc->buffer_time = DEFAULT_BUFFER_TIME;
baseaudiosrc->latency_time = DEFAULT_LATENCY_TIME;
+ /* reset blocksize we use latency time to calculate a more useful
+ * value based on negotiated format. */
+ GST_BASE_SRC (baseaudiosrc)->blocksize = 0;
baseaudiosrc->clock = gst_audio_clock_new ("clock",
(GstAudioClockGetTimeFunc) gst_base_audio_src_get_time, baseaudiosrc);
gst_pad_set_fixatecaps_function (GST_BASE_SRC_PAD (baseaudiosrc),
gst_base_audio_src_fixate);
+ /* we are always a live source */
gst_base_src_set_live (GST_BASE_SRC (baseaudiosrc), TRUE);
gst_base_src_set_format (GST_BASE_SRC (baseaudiosrc), GST_FORMAT_TIME);
}
+static gboolean
+gst_base_audio_src_set_clock (GstElement * elem, GstClock * clock)
+{
+ GstBaseAudioSrc *src;
+
+ src = GST_BASE_AUDIO_SRC (elem);
+
+ /* FIXME, we cannot slave to another clock yet, better fail
+ * than to give a bad user experience (tm). */
+ if (clock && clock != src->clock)
+ goto wrong_clock;
+
+ return TRUE;
+
+ /* ERRORS */
+wrong_clock:
+ {
+ GST_ELEMENT_ERROR (src, CORE, CLOCK,
+ (NULL), ("Cannot operate with this clock."));
+ return FALSE;
+ }
+}
+
static GstClock *
gst_base_audio_src_provide_clock (GstElement * elem)
{
GstBaseAudioSrc *src;
+ GstClock *clock;
src = GST_BASE_AUDIO_SRC (elem);
- return GST_CLOCK (gst_object_ref (GST_OBJECT (src->clock)));
+ /* we have no ringbuffer (must be NULL state) */
+ if (src->ringbuffer == NULL)
+ goto wrong_state;
+
+ if (!gst_ring_buffer_is_acquired (src->ringbuffer))
+ goto wrong_state;
+
+ clock = GST_CLOCK_CAST (gst_object_ref (src->clock));
+
+ return clock;
+
+ /* ERRORS */
+wrong_state:
+ {
+ GST_DEBUG_OBJECT (src, "ringbuffer not acquired");
+ return NULL;
+ }
}
static GstClockTime
guint64 samples;
GstClockTime result;
- if (src->ringbuffer == NULL || src->ringbuffer->spec.rate == 0)
- return 0;
+ if (G_UNLIKELY (src->ringbuffer == NULL || src->ringbuffer->spec.rate == 0))
+ return GST_CLOCK_TIME_NONE;
samples = gst_ring_buffer_samples_done (src->ringbuffer);
return result;
}
+static gboolean
+gst_base_audio_src_check_get_range (GstBaseSrc * bsrc)
+{
+ /* we allow limited pull base operation of which the details
+ * will eventually exposed in an as of yet non-existing query.
+ * Basically pulling can be done on any number of bytes as long
+ * as the offset is -1 or sequentially increasing. */
+ return TRUE;
+}
+
static void
gst_base_audio_src_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
}
static GstFlowReturn
-gst_base_audio_src_create (GstPushSrc * psrc, GstBuffer ** outbuf)
+gst_base_audio_src_create (GstBaseSrc * bsrc, guint64 offset, guint length,
+ GstBuffer ** outbuf)
{
- GstBaseAudioSrc *src = GST_BASE_AUDIO_SRC (psrc);
+ GstBaseAudioSrc *src = GST_BASE_AUDIO_SRC (bsrc);
GstBuffer *buf;
guchar *data;
- guint len, samples;
+ guint samples;
guint res;
guint64 sample;
+ gint bps;
GstRingBuffer *ringbuffer;
ringbuffer = src->ringbuffer;
- if (!gst_ring_buffer_is_acquired (ringbuffer))
+ if (G_UNLIKELY (!gst_ring_buffer_is_acquired (ringbuffer)))
goto wrong_state;
- buf = gst_buffer_new_and_alloc (ringbuffer->spec.segsize);
+ bps = ringbuffer->spec.bytes_per_sample;
- data = GST_BUFFER_DATA (buf);
- len = GST_BUFFER_SIZE (buf);
+ if ((length == 0 && bsrc->blocksize == 0) || length == -1)
+ /* no length given, use the default segment size */
+ length = ringbuffer->spec.segsize;
+ else
+ /* make sure we round down to an integral number of samples */
+ length -= length % bps;
- if (src->next_sample != -1) {
- sample = src->next_sample;
- } else {
- sample = 0;
+ /* calculate the sequentially next sample we need to read */
+ sample = (src->next_sample != -1 ? src->next_sample : 0);
+
+ if (G_UNLIKELY (offset != -1)) {
+ /* if a specific offset was given it must be the next
+ * sequential offset we expect or we fail. */
+ if (offset / bps != sample)
+ goto wrong_offset;
}
- samples = len / ringbuffer->spec.bytes_per_sample;
+ /* get the number of samples to read */
+ samples = length / bps;
+
+ /* FIXME, using a bufferpool would be nice here */
+ buf = gst_buffer_new_and_alloc (length);
+ data = GST_BUFFER_DATA (buf);
res = gst_ring_buffer_read (ringbuffer, sample, data, samples);
- if (res == -1)
+ if (G_UNLIKELY (res == -1))
goto stopped;
+ /* FIXME, we timestamp against our own clock, also handle the case
+ * where we are slaved to another clock. We currently refuse to accept
+ * any other clock than the one we provide, so this code is fine for
+ * now. */
GST_BUFFER_TIMESTAMP (buf) = gst_util_uint64_scale_int (sample,
GST_SECOND, ringbuffer->spec.rate);
src->next_sample = sample + samples;
GST_BUFFER_DURATION (buf) = gst_util_uint64_scale_int (src->next_sample,
GST_SECOND, ringbuffer->spec.rate) - GST_BUFFER_TIMESTAMP (buf);
- gst_buffer_set_caps (buf, GST_PAD_CAPS (GST_BASE_SRC_PAD (psrc)));
+ gst_buffer_set_caps (buf, GST_PAD_CAPS (GST_BASE_SRC_PAD (bsrc)));
*outbuf = buf;
return GST_FLOW_OK;
+ /* ERRORS */
wrong_state:
{
- GST_DEBUG ("ringbuffer in wrong state");
+ GST_DEBUG_OBJECT (src, "ringbuffer in wrong state");
return GST_FLOW_WRONG_STATE;
}
+wrong_offset:
+ {
+ GST_ELEMENT_ERROR (src, RESOURCE, SEEK,
+ (NULL), ("resource can only be operated on sequentially but offset %"
+ G_GUINT64_FORMAT " was given", offset));
+ return GST_FLOW_ERROR;
+ }
stopped:
{
gst_buffer_unref (buf);
- GST_DEBUG ("ringbuffer stopped");
+ GST_DEBUG_OBJECT (src, "ringbuffer stopped");
return GST_FLOW_WRONG_STATE;
}
}
if (bclass->create_ringbuffer)
buffer = bclass->create_ringbuffer (src);
- if (buffer) {
- gst_object_set_parent (GST_OBJECT (buffer), GST_OBJECT (src));
- }
+ if (G_LIKELY (buffer))
+ gst_object_set_parent (GST_OBJECT_CAST (buffer), GST_OBJECT_CAST (src));
return buffer;
}
projects / platform / upstream / gstreamer.git / commitdiff