/**
* SECTION:element-audioinvert
- * @short_description: Swaps upper and lower half of audio samples
*
- * <refsect2>
* Swaps upper and lower half of audio samples. Mixing an inverted sample on top of
* the original with a slight delay can produce effects that sound like resonance.
* Creating a stereo sample from a mono source, with one channel inverted produces wide-stereo sounds.
+ *
+ * <refsect2>
* <title>Example launch line</title>
- * <para>
- * <programlisting>
+ * |[
* gst-launch audiotestsrc wave=saw ! audioinvert invert=0.4 ! alsasink
* gst-launch filesrc location="melo1.ogg" ! oggdemux ! vorbisdec ! audioconvert ! audioinvert invert=0.4 ! alsasink
* gst-launch audiotestsrc wave=saw ! audioconvert ! audioinvert invert=0.4 ! audioconvert ! alsasink
- * </programlisting>
- * </para>
+ * ]|
* </refsect2>
*/
#include <gst/base/gstbasetransform.h>
#include <gst/audio/audio.h>
#include <gst/audio/gstaudiofilter.h>
-#include <gst/controller/gstcontroller.h>
#include "audioinvert.h"
#define GST_CAT_DEFAULT gst_audio_invert_debug
GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
-static const GstElementDetails element_details =
-GST_ELEMENT_DETAILS ("Audio inversion",
- "Filter/Effect/Audio",
- "Swaps upper and lower half of audio samples",
- "Sebastian Dröge <slomo@circular-chaos.org>");
-
/* Filter signals and args */
enum
{
};
#define ALLOWED_CAPS \
- "audio/x-raw-int," \
- " depth=(int)16," \
- " width=(int)16," \
- " endianness=(int)BYTE_ORDER," \
- " signed=(bool)TRUE," \
- " rate=(int)[1,MAX]," \
- " channels=(int)[1,MAX]; " \
- "audio/x-raw-float," \
- " width=(int)32," \
- " endianness=(int)BYTE_ORDER," \
- " rate=(int)[1,MAX]," \
- " channels=(int)[1,MAX]"
-
-#define DEBUG_INIT(bla) \
- GST_DEBUG_CATEGORY_INIT (gst_audio_invert_debug, "audioinvert", 0, "audioinvert element");
-
-GST_BOILERPLATE_FULL (GstAudioInvert, gst_audio_invert, GstAudioFilter,
- GST_TYPE_AUDIO_FILTER, DEBUG_INIT);
+ "audio/x-raw," \
+ " format=(string) {"GST_AUDIO_NE(S16)","GST_AUDIO_NE(F32)"}," \
+ " rate=(int)[1,MAX]," \
+ " channels=(int)[1,MAX] "
+
+G_DEFINE_TYPE (GstAudioInvert, gst_audio_invert, GST_TYPE_AUDIO_FILTER);
static void gst_audio_invert_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
GValue * value, GParamSpec * pspec);
static gboolean gst_audio_invert_setup (GstAudioFilter * filter,
- GstRingBufferSpec * format);
+ const GstAudioInfo * info);
static GstFlowReturn gst_audio_invert_transform_ip (GstBaseTransform * base,
GstBuffer * buf);
/* GObject vmethod implementations */
static void
-gst_audio_invert_base_init (gpointer klass)
-{
- GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
- GstCaps *caps;
-
- gst_element_class_set_details (element_class, &element_details);
-
- caps = gst_caps_from_string (ALLOWED_CAPS);
- gst_audio_filter_class_add_pad_templates (GST_AUDIO_FILTER_CLASS (klass),
- caps);
- gst_caps_unref (caps);
-}
-
-static void
gst_audio_invert_class_init (GstAudioInvertClass * klass)
{
GObjectClass *gobject_class;
+ GstElementClass *gstelement_class;
+ GstCaps *caps;
+
+ GST_DEBUG_CATEGORY_INIT (gst_audio_invert_debug, "audioinvert", 0,
+ "audioinvert element");
gobject_class = (GObjectClass *) klass;
+ gstelement_class = (GstElementClass *) klass;
+
gobject_class->set_property = gst_audio_invert_set_property;
gobject_class->get_property = gst_audio_invert_get_property;
g_object_class_install_property (gobject_class, PROP_DEGREE,
g_param_spec_float ("degree", "Degree",
"Degree of inversion", 0.0, 1.0,
- 0.0, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
+ 0.0,
+ G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE | G_PARAM_STATIC_STRINGS));
+
+ gst_element_class_set_details_simple (gstelement_class, "Audio inversion",
+ "Filter/Effect/Audio",
+ "Swaps upper and lower half of audio samples",
+ "Sebastian Dröge <slomo@circular-chaos.org>");
+
+ caps = gst_caps_from_string (ALLOWED_CAPS);
+ gst_audio_filter_class_add_pad_templates (GST_AUDIO_FILTER_CLASS (klass),
+ caps);
+ gst_caps_unref (caps);
GST_AUDIO_FILTER_CLASS (klass)->setup =
GST_DEBUG_FUNCPTR (gst_audio_invert_setup);
}
static void
-gst_audio_invert_init (GstAudioInvert * filter, GstAudioInvertClass * klass)
+gst_audio_invert_init (GstAudioInvert * filter)
{
filter->degree = 0.0;
gst_base_transform_set_in_place (GST_BASE_TRANSFORM (filter), TRUE);
/* GstAudioFilter vmethod implementations */
static gboolean
-gst_audio_invert_setup (GstAudioFilter * base, GstRingBufferSpec * format)
+gst_audio_invert_setup (GstAudioFilter * base, const GstAudioInfo * info)
{
GstAudioInvert *filter = GST_AUDIO_INVERT (base);
gboolean ret = TRUE;
- if (format->type == GST_BUFTYPE_FLOAT && format->width == 32)
- filter->process = (GstAudioInvertProcessFunc)
- gst_audio_invert_transform_float;
- else if (format->type == GST_BUFTYPE_LINEAR && format->width == 16)
- filter->process = (GstAudioInvertProcessFunc)
- gst_audio_invert_transform_int;
- else
- ret = FALSE;
-
+ switch (GST_AUDIO_INFO_FORMAT (info)) {
+ case GST_AUDIO_FORMAT_S16:
+ filter->process = (GstAudioInvertProcessFunc)
+ gst_audio_invert_transform_int;
+ break;
+ case GST_AUDIO_FORMAT_F32:
+ filter->process = (GstAudioInvertProcessFunc)
+ gst_audio_invert_transform_float;
+ break;
+ default:
+ ret = FALSE;
+ break;
+ }
return ret;
}
gst_audio_invert_transform_ip (GstBaseTransform * base, GstBuffer * buf)
{
GstAudioInvert *filter = GST_AUDIO_INVERT (base);
- guint num_samples =
- GST_BUFFER_SIZE (buf) / (GST_AUDIO_FILTER (filter)->format.width / 8);
+ guint num_samples;
+ GstClockTime timestamp, stream_time;
+ guint8 *data;
+ gsize size;
- if (GST_CLOCK_TIME_IS_VALID (GST_BUFFER_TIMESTAMP (buf)))
- gst_object_sync_values (G_OBJECT (filter), GST_BUFFER_TIMESTAMP (buf));
+ timestamp = GST_BUFFER_TIMESTAMP (buf);
+ stream_time =
+ gst_segment_to_stream_time (&base->segment, GST_FORMAT_TIME, timestamp);
+
+ GST_DEBUG_OBJECT (filter, "sync to %" GST_TIME_FORMAT,
+ GST_TIME_ARGS (timestamp));
+
+ if (GST_CLOCK_TIME_IS_VALID (stream_time))
+ gst_object_sync_values (GST_OBJECT (filter), stream_time);
if (gst_base_transform_is_passthrough (base) ||
G_UNLIKELY (GST_BUFFER_FLAG_IS_SET (buf, GST_BUFFER_FLAG_GAP)))
return GST_FLOW_OK;
- filter->process (filter, GST_BUFFER_DATA (buf), num_samples);
+ data = gst_buffer_map (buf, &size, NULL, GST_MAP_READWRITE);
+ num_samples = size / GST_AUDIO_FILTER_BPS (filter);
+
+ filter->process (filter, data, num_samples);
+
+ gst_buffer_unmap (buf, data, size);
return GST_FLOW_OK;
}
projects / platform / upstream / gst-plugins-good.git / blobdiff