Commit 7df4af3e authored by Jan Schmidt's avatar Jan Schmidt

Use bytestream in goom for input samples

Original commit message from CVS:
Use bytestream in goom for input samples
Add a debug category
parent fcc488c8
2004-11-11 Jan Schmidt <thaytan@mad.scientist.com>
* gst/goom/gstgoom.c: (gst_goom_class_init), (gst_goom_init),
(gst_goom_dispose), (gst_goom_sinkconnect), (gst_goom_chain),
(gst_goom_change_state), (plugin_init):
Use the bytestream adapter so goom doesn't depend on the input
buffer size.
Add a debug category
2004-11-11 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
* ext/alsa/gstalsa.c: (gst_alsa_change_state):
......
......@@ -21,10 +21,17 @@
#include "config.h"
#endif
#include <string.h>
#include <gst/gst.h>
#include <gst/video/video.h>
#include <gst/bytestream/adapter.h>
#include "goom_core.h"
GST_DEBUG_CATEGORY_STATIC (goom_debug);
#define GST_CAT_DEFAULT goom_debug
#define GOOM_SAMPLES 512
#define GST_TYPE_GOOM (gst_goom_get_type())
#define GST_GOOM(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_GOOM,GstGOOM))
#define GST_GOOM_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_GOOM,GstGOOM))
......@@ -40,10 +47,15 @@ struct _GstGOOM
/* pads */
GstPad *sinkpad, *srcpad;
GstAdapter *adapter;
/* input tracking */
gint sample_rate;
gint16 datain[2][GOOM_SAMPLES];
/* the timestamp of the next frame */
guint64 next_time;
gint16 datain[2][512];
GstClockTime audio_basetime;
guint64 samples_consumed;
/* video state */
gdouble fps;
......@@ -51,6 +63,8 @@ struct _GstGOOM
gint height;
gint channels;
gboolean srcnegotiated;
gboolean disposed;
};
struct _GstGOOMClass
......@@ -166,6 +180,8 @@ gst_goom_class_init (GstGOOMClass * klass)
gobject_class->dispose = gst_goom_dispose;
gstelement_class->change_state = gst_goom_change_state;
GST_DEBUG_CATEGORY_INIT (goom_debug, "goom", 0, "goom visualisation element");
}
static void
......@@ -189,10 +205,17 @@ gst_goom_init (GstGOOM * goom)
gst_pad_set_link_function (goom->srcpad, gst_goom_srcconnect);
gst_pad_set_fixate_function (goom->srcpad, gst_goom_src_fixate);
goom->adapter = gst_adapter_new ();
goom->width = 320;
goom->height = 200;
goom->fps = 25.; /* desired frame rate */
goom->channels = 0;
goom->sample_rate = 0;
goom->audio_basetime = GST_CLOCK_TIME_NONE;
goom->samples_consumed = 0;
goom->disposed = FALSE;
/* set to something */
goom_init (50, 50);
}
......@@ -200,7 +223,15 @@ gst_goom_init (GstGOOM * goom)
static void
gst_goom_dispose (GObject * object)
{
goom_close ();
GstGOOM *goom = GST_GOOM (object);
if (!goom->disposed) {
goom_close ();
goom->disposed = TRUE;
g_object_unref (goom->adapter);
goom->adapter = NULL;
}
G_OBJECT_CLASS (parent_class)->dispose (object);
}
......@@ -216,7 +247,7 @@ gst_goom_sinkconnect (GstPad * pad, const GstCaps * caps)
structure = gst_caps_get_structure (caps, 0);
gst_structure_get_int (structure, "channels", &goom->channels);
gst_structure_get_int (structure, "rate", &goom->sample_rate);
return GST_PAD_LINK_OK;
}
......@@ -273,15 +304,11 @@ gst_goom_chain (GstPad * pad, GstData * _data)
{
GstBuffer *bufin = GST_BUFFER (_data);
GstGOOM *goom;
GstBuffer *bufout;
guint32 samples_in;
guint32 bytesperread;
gint16 *data;
gint i;
gint samples_per_frame;
goom = GST_GOOM (gst_pad_get_parent (pad));
GST_DEBUG ("GOOM: chainfunc called");
if (GST_IS_EVENT (bufin)) {
GstEvent *event = GST_EVENT (bufin);
......@@ -291,8 +318,10 @@ gst_goom_chain (GstPad * pad, GstData * _data)
gint64 value = 0;
gst_event_discont_get_value (event, GST_FORMAT_TIME, &value);
goom->next_time = value;
gst_adapter_clear (goom->adapter);
goom->audio_basetime = value;
goom->samples_consumed = 0;
GST_DEBUG ("Got discont. Adjusting time to=%" G_GUINT64_FORMAT, value);
}
default:
gst_pad_event_default (pad, event);
......@@ -303,49 +332,73 @@ gst_goom_chain (GstPad * pad, GstData * _data)
if (goom->channels == 0) {
GST_ELEMENT_ERROR (goom, CORE, NEGOTIATION, (NULL),
("format wasn't negotiated before chain function"));
goto done;
("Format wasn't negotiated before chain function"));
gst_buffer_unref (bufin);
return;
}
if (!GST_PAD_IS_USABLE (goom->srcpad))
goto done;
samples_in = GST_BUFFER_SIZE (bufin) / (sizeof (gint16) * goom->channels);
if (!GST_PAD_IS_USABLE (goom->srcpad)) {
gst_buffer_unref (bufin);
return;
}
GST_DEBUG ("input buffer has %d samples", samples_in);
if (goom->audio_basetime == GST_CLOCK_TIME_NONE)
goom->audio_basetime = GST_BUFFER_TIMESTAMP (bufin);
if (GST_BUFFER_TIMESTAMP (bufin) < goom->next_time || samples_in < 512) {
goto done;
}
if (goom->audio_basetime == GST_CLOCK_TIME_NONE)
goom->audio_basetime = 0;
bytesperread = GOOM_SAMPLES * goom->channels * sizeof (gint16);
samples_per_frame = goom->sample_rate / goom->fps;
data = (gint16 *) GST_BUFFER_DATA (bufin);
if (goom->channels == 2) {
for (i = 0; i < 512; i++) {
goom->datain[0][i] = *data++;
goom->datain[1][i] = *data++;
}
} else {
for (i = 0; i < 512; i++) {
goom->datain[0][i] = *data;
goom->datain[1][i] = *data++;
}
}
bufout = gst_buffer_new ();
GST_BUFFER_SIZE (bufout) = goom->width * goom->height * 4;
GST_BUFFER_DATA (bufout) = (guchar *) goom_update (goom->datain);
GST_BUFFER_TIMESTAMP (bufout) = goom->next_time;
GST_BUFFER_FLAG_SET (bufout, GST_BUFFER_DONTFREE);
gst_adapter_push (goom->adapter, bufin);
GST_DEBUG ("Input buffer has %d samples, time=%" G_GUINT64_FORMAT,
GST_BUFFER_SIZE (bufin) * sizeof (gint16) * goom->channels,
GST_BUFFER_TIMESTAMP (bufin));
/* Collect samples until we have enough for an output frame */
while (gst_adapter_available (goom->adapter) > MAX (bytesperread,
samples_per_frame * goom->channels * sizeof (gint16))) {
const guint16 *data =
(const guint16 *) gst_adapter_peek (goom->adapter, bytesperread);
GstBuffer *bufout;
guchar *out_frame;
GstClockTimeDiff frame_duration = GST_SECOND / goom->fps;
gint i;
if (goom->channels == 2) {
for (i = 0; i < GOOM_SAMPLES; i++) {
goom->datain[0][i] = *data++;
goom->datain[1][i] = *data++;
}
} else {
for (i = 0; i < GOOM_SAMPLES; i++) {
goom->datain[0][i] = *data;
goom->datain[1][i] = *data++;
}
}
goom->next_time += GST_SECOND / goom->fps;
bufout = gst_buffer_new_and_alloc (goom->width * goom->height * 4);
GST_BUFFER_TIMESTAMP (bufout) =
goom->audio_basetime +
(GST_SECOND * goom->samples_consumed / goom->sample_rate);
GST_BUFFER_DURATION (bufout) = frame_duration;
GST_BUFFER_SIZE (bufout) = goom->width * goom->height * 4;
gst_pad_push (goom->srcpad, GST_DATA (bufout));
out_frame = (guchar *) goom_update (goom->datain);
memcpy (GST_BUFFER_DATA (bufout), out_frame, GST_BUFFER_SIZE (bufout));
done:
gst_buffer_unref (bufin);
GST_DEBUG ("Pushing frame with time=%" G_GUINT64_FORMAT ", duration=%"
G_GUINT64_FORMAT, GST_BUFFER_TIMESTAMP (bufout),
GST_BUFFER_DURATION (bufout));
gst_pad_push (goom->srcpad, GST_DATA (bufout));
GST_DEBUG ("GOOM: exiting chainfunc");
goom->samples_consumed += samples_per_frame;
gst_adapter_flush (goom->adapter,
samples_per_frame * goom->channels * sizeof (gint16));
}
}
static GstElementStateReturn
......@@ -359,8 +412,9 @@ gst_goom_change_state (GstElement * element)
case GST_STATE_READY_TO_NULL:
break;
case GST_STATE_READY_TO_PAUSED:
goom->next_time = 0;
goom->audio_basetime = GST_CLOCK_TIME_NONE;
goom->srcnegotiated = FALSE;
gst_adapter_clear (goom->adapter);
break;
case GST_STATE_PAUSED_TO_READY:
goom->channels = 0;
......@@ -378,6 +432,8 @@ gst_goom_change_state (GstElement * element)
static gboolean
plugin_init (GstPlugin * plugin)
{
if (!gst_library_load ("gstbytestream"))
return FALSE;
return gst_element_register (plugin, "goom", GST_RANK_NONE, GST_TYPE_GOOM);
}
......
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