--- /dev/null Thu Jan 01 00:00:00 1970 +0000
+++ b/gst_plugins_good/gst/audiofx/audioecho.c Wed Mar 24 18:04:17 2010 -0500
@@ -0,0 +1,391 @@
+/*
+ * GStreamer
+ * Copyright (C) 2009 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
+ * Boston, MA 02111-1307, USA.
+ */
+
+/**
+ * SECTION:element-audioecho
+ * @Since: 0.10.14
+ *
+ * audioecho adds an echo or (simple) reverb effect to an audio stream. The echo
+ * delay, intensity and the percentage of feedback can be configured.
+ *
+ * For getting an echo effect you have to set the delay to a larger value,
+ * for example 200ms and more. Everything below will result in a simple
+ * reverb effect, which results in a slightly metallic sound.
+ *
+ * Use the max-delay property to set the maximum amount of delay that
+ * will be used. This can only be set before going to the PAUSED or PLAYING
+ * state and will be set to the current delay by default.
+ *
+ * <refsect2>
+ * <title>Example launch line</title>
+ * |[
+ * gst-launch filesrc location="melo1.ogg" ! audioconvert ! audioecho delay=500000000 intensity=0.6 feedback=0.4 ! audioconvert ! autoaudiosink
+ * gst-launch filesrc location="melo1.ogg" ! decodebin ! audioconvert ! audioecho delay=50000000 intensity=0.6 feedback=0.4 ! audioconvert ! autoaudiosink
+ * ]|
+ * </refsect2>
+ */
+
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+
+#include <gst/gst.h>
+#include <gst/base/gstbasetransform.h>
+#include <gst/audio/audio.h>
+#include <gst/audio/gstaudiofilter.h>
+#include <gst/controller/gstcontroller.h>
+
+#include "audioecho.h"
+
+#define GST_CAT_DEFAULT gst_audio_echo_debug
+GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
+
+enum
+{
+ PROP_0,
+ PROP_DELAY,
+ PROP_MAX_DELAY,
+ PROP_INTENSITY,
+ PROP_FEEDBACK
+};
+
+#define ALLOWED_CAPS \
+ "audio/x-raw-float," \
+ " width=(int) { 32, 64 }, " \
+ " endianness=(int)BYTE_ORDER," \
+ " rate=(int)[1,MAX]," \
+ " channels=(int)[1,MAX]"
+
+#define DEBUG_INIT(bla) \
+ GST_DEBUG_CATEGORY_INIT (gst_audio_echo_debug, "audioecho", 0, "audioecho element");
+
+GST_BOILERPLATE_FULL (GstAudioEcho, gst_audio_echo, GstAudioFilter,
+ GST_TYPE_AUDIO_FILTER, DEBUG_INIT);
+
+static void gst_audio_echo_set_property (GObject * object, guint prop_id,
+ const GValue * value, GParamSpec * pspec);
+static void gst_audio_echo_get_property (GObject * object, guint prop_id,
+ GValue * value, GParamSpec * pspec);
+static void gst_audio_echo_finalize (GObject * object);
+
+static gboolean gst_audio_echo_setup (GstAudioFilter * self,
+ GstRingBufferSpec * format);
+static gboolean gst_audio_echo_stop (GstBaseTransform * base);
+static GstFlowReturn gst_audio_echo_transform_ip (GstBaseTransform * base,
+ GstBuffer * buf);
+
+static void gst_audio_echo_transform_float (GstAudioEcho * self,
+ gfloat * data, guint num_samples);
+static void gst_audio_echo_transform_double (GstAudioEcho * self,
+ gdouble * data, guint num_samples);
+
+/* GObject vmethod implementations */
+
+static void
+gst_audio_echo_base_init (gpointer klass)
+{
+ GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
+ GstCaps *caps;
+
+ gst_element_class_set_details_simple (element_class, "Audio echo",
+ "Filter/Effect/Audio",
+ "Adds an echo or reverb effect to an audio stream",
+ "Sebastian Dröge <sebastian.droege@collabora.co.uk>");
+
+ 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_echo_class_init (GstAudioEchoClass * klass)
+{
+ GObjectClass *gobject_class = (GObjectClass *) klass;
+ GstBaseTransformClass *basetransform_class = (GstBaseTransformClass *) klass;
+ GstAudioFilterClass *audioself_class = (GstAudioFilterClass *) klass;
+
+ gobject_class->set_property = gst_audio_echo_set_property;
+ gobject_class->get_property = gst_audio_echo_get_property;
+ gobject_class->finalize = gst_audio_echo_finalize;
+
+ g_object_class_install_property (gobject_class, PROP_DELAY,
+ g_param_spec_uint64 ("delay", "Delay",
+ "Delay of the echo in nanoseconds", 1, G_MAXUINT64,
+ 1, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS
+ | GST_PARAM_CONTROLLABLE));
+
+ g_object_class_install_property (gobject_class, PROP_MAX_DELAY,
+ g_param_spec_uint64 ("max-delay", "Maximum Delay",
+ "Maximum delay of the echo in nanoseconds"
+ " (can't be changed in PLAYING or PAUSED state)",
+ 1, G_MAXUINT64, 1,
+ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS | GST_PARAM_CONTROLLABLE));
+
+ g_object_class_install_property (gobject_class, PROP_INTENSITY,
+ g_param_spec_float ("intensity", "Intensity",
+ "Intensity of the echo", 0.0, 1.0,
+ 0.0, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS
+ | GST_PARAM_CONTROLLABLE));
+
+ g_object_class_install_property (gobject_class, PROP_FEEDBACK,
+ g_param_spec_float ("feedback", "Feedback",
+ "Amount of feedback", 0.0, 1.0,
+ 0.0, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS
+ | GST_PARAM_CONTROLLABLE));
+
+ audioself_class->setup = GST_DEBUG_FUNCPTR (gst_audio_echo_setup);
+ basetransform_class->transform_ip =
+ GST_DEBUG_FUNCPTR (gst_audio_echo_transform_ip);
+ basetransform_class->stop = GST_DEBUG_FUNCPTR (gst_audio_echo_stop);
+}
+
+static void
+gst_audio_echo_init (GstAudioEcho * self, GstAudioEchoClass * klass)
+{
+ self->delay = 1;
+ self->max_delay = 1;
+ self->intensity = 0.0;
+ self->feedback = 0.0;
+
+ gst_base_transform_set_in_place (GST_BASE_TRANSFORM (self), TRUE);
+}
+
+static void
+gst_audio_echo_finalize (GObject * object)
+{
+ GstAudioEcho *self = GST_AUDIO_ECHO (object);
+
+ g_free (self->buffer);
+ self->buffer = NULL;
+
+ G_OBJECT_CLASS (parent_class)->finalize (object);
+}
+
+static void
+gst_audio_echo_set_property (GObject * object, guint prop_id,
+ const GValue * value, GParamSpec * pspec)
+{
+ GstAudioEcho *self = GST_AUDIO_ECHO (object);
+
+ switch (prop_id) {
+ case PROP_DELAY:{
+ guint64 max_delay, delay;
+
+ GST_BASE_TRANSFORM_LOCK (self);
+ delay = g_value_get_uint64 (value);
+ max_delay = self->max_delay;
+
+ if (delay > max_delay && GST_STATE (self) > GST_STATE_READY) {
+ GST_WARNING_OBJECT (self, "New delay (%" GST_TIME_FORMAT ") "
+ "is larger than maximum delay (%" GST_TIME_FORMAT ")",
+ GST_TIME_ARGS (delay), GST_TIME_ARGS (max_delay));
+ self->delay = max_delay;
+ } else {
+ self->delay = delay;
+ self->max_delay = MAX (delay, max_delay);
+ }
+ GST_BASE_TRANSFORM_UNLOCK (self);
+ }
+ break;
+ case PROP_MAX_DELAY:{
+ guint64 max_delay, delay;
+
+ GST_BASE_TRANSFORM_LOCK (self);
+ max_delay = g_value_get_uint64 (value);
+ delay = self->delay;
+
+ if (GST_STATE (self) > GST_STATE_READY) {
+ GST_ERROR_OBJECT (self, "Can't change maximum delay in"
+ " PLAYING or PAUSED state");
+ } else {
+ self->delay = delay;
+ self->max_delay = max_delay;
+ }
+ GST_BASE_TRANSFORM_UNLOCK (self);
+ }
+ break;
+ case PROP_INTENSITY:{
+ GST_BASE_TRANSFORM_LOCK (self);
+ self->intensity = g_value_get_float (value);
+ GST_BASE_TRANSFORM_UNLOCK (self);
+ }
+ break;
+ case PROP_FEEDBACK:{
+ GST_BASE_TRANSFORM_LOCK (self);
+ self->feedback = g_value_get_float (value);
+ GST_BASE_TRANSFORM_UNLOCK (self);
+ }
+ break;
+ default:
+ G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+ break;
+ }
+}
+
+static void
+gst_audio_echo_get_property (GObject * object, guint prop_id,
+ GValue * value, GParamSpec * pspec)
+{
+ GstAudioEcho *self = GST_AUDIO_ECHO (object);
+
+ switch (prop_id) {
+ case PROP_DELAY:
+ GST_BASE_TRANSFORM_LOCK (self);
+ g_value_set_uint64 (value, self->delay);
+ GST_BASE_TRANSFORM_UNLOCK (self);
+ break;
+ case PROP_MAX_DELAY:
+ GST_BASE_TRANSFORM_LOCK (self);
+ g_value_set_uint64 (value, self->max_delay);
+ GST_BASE_TRANSFORM_UNLOCK (self);
+ break;
+ case PROP_INTENSITY:
+ GST_BASE_TRANSFORM_LOCK (self);
+ g_value_set_float (value, self->intensity);
+ GST_BASE_TRANSFORM_UNLOCK (self);
+ break;
+ case PROP_FEEDBACK:
+ GST_BASE_TRANSFORM_LOCK (self);
+ g_value_set_float (value, self->feedback);
+ GST_BASE_TRANSFORM_UNLOCK (self);
+ break;
+ default:
+ G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+ break;
+ }
+}
+
+/* GstAudioFilter vmethod implementations */
+
+static gboolean
+gst_audio_echo_setup (GstAudioFilter * base, GstRingBufferSpec * format)
+{
+ GstAudioEcho *self = GST_AUDIO_ECHO (base);
+ gboolean ret = TRUE;
+
+ if (format->type == GST_BUFTYPE_FLOAT && format->width == 32)
+ self->process = (GstAudioEchoProcessFunc)
+ gst_audio_echo_transform_float;
+ else if (format->type == GST_BUFTYPE_FLOAT && format->width == 64)
+ self->process = (GstAudioEchoProcessFunc)
+ gst_audio_echo_transform_double;
+ else
+ ret = FALSE;
+
+ g_free (self->buffer);
+ self->buffer = NULL;
+ self->buffer_pos = 0;
+ self->buffer_size = 0;
+ self->buffer_size_frames = 0;
+
+ return ret;
+}
+
+static gboolean
+gst_audio_echo_stop (GstBaseTransform * base)
+{
+ GstAudioEcho *self = GST_AUDIO_ECHO (base);
+
+ g_free (self->buffer);
+ self->buffer = NULL;
+ self->buffer_pos = 0;
+ self->buffer_size = 0;
+ self->buffer_size_frames = 0;
+
+ return TRUE;
+}
+
+#define TRANSFORM_FUNC(name, type) \
+static void \
+gst_audio_echo_transform_##name (GstAudioEcho * self, \
+ type * data, guint num_samples) \
+{ \
+ type *buffer = (type *) self->buffer; \
+ guint channels = GST_AUDIO_FILTER (self)->format.channels; \
+ guint rate = GST_AUDIO_FILTER (self)->format.rate; \
+ guint i, j; \
+ guint echo_index = self->buffer_size_frames - self->delay_frames; \
+ gdouble echo_off = ((((gdouble) self->delay) * rate) / GST_SECOND) - self->delay_frames; \
+ \
+ if (echo_off < 0.0) \
+ echo_off = 0.0; \
+ \
+ num_samples /= channels; \
+ \
+ for (i = 0; i < num_samples; i++) { \
+ guint echo0_index = ((echo_index + self->buffer_pos) % self->buffer_size_frames) * channels; \
+ guint echo1_index = ((echo_index + self->buffer_pos +1) % self->buffer_size_frames) * channels; \
+ guint rbout_index = (self->buffer_pos % self->buffer_size_frames) * channels; \
+ for (j = 0; j < channels; j++) { \
+ gdouble in = data[i*channels + j]; \
+ gdouble echo0 = buffer[echo0_index + j]; \
+ gdouble echo1 = buffer[echo1_index + j]; \
+ gdouble echo = echo0 + (echo1-echo0)*echo_off; \
+ type out = in + self->intensity * echo; \
+ \
+ data[i*channels + j] = out; \
+ \
+ buffer[rbout_index + j] = in + self->feedback * echo; \
+ } \
+ self->buffer_pos = (self->buffer_pos + 1) % self->buffer_size_frames; \
+ } \
+}
+
+TRANSFORM_FUNC (float, gfloat);
+TRANSFORM_FUNC (double, gdouble);
+
+/* GstBaseTransform vmethod implementations */
+static GstFlowReturn
+gst_audio_echo_transform_ip (GstBaseTransform * base, GstBuffer * buf)
+{
+ GstAudioEcho *self = GST_AUDIO_ECHO (base);
+ guint num_samples =
+ GST_BUFFER_SIZE (buf) / (GST_AUDIO_FILTER (self)->format.width / 8);
+
+ if (GST_CLOCK_TIME_IS_VALID (GST_BUFFER_TIMESTAMP (buf)))
+ gst_object_sync_values (G_OBJECT (self), GST_BUFFER_TIMESTAMP (buf));
+
+ if (self->buffer == NULL) {
+ guint width, rate, channels;
+
+ width = GST_AUDIO_FILTER (self)->format.width / 8;
+ rate = GST_AUDIO_FILTER (self)->format.rate;
+ channels = GST_AUDIO_FILTER (self)->format.channels;
+
+ self->delay_frames =
+ MAX (gst_util_uint64_scale (self->delay, rate, GST_SECOND), 1);
+ self->buffer_size_frames =
+ MAX (gst_util_uint64_scale (self->max_delay, rate, GST_SECOND), 1);
+
+ self->buffer_size = self->buffer_size_frames * width * channels;
+ self->buffer = g_try_malloc0 (self->buffer_size);
+ self->buffer_pos = 0;
+
+ if (self->buffer == NULL) {
+ GST_ERROR_OBJECT (self, "Failed to allocate %u bytes", self->buffer_size);
+ return GST_FLOW_ERROR;
+ }
+ }
+
+ self->process (self, GST_BUFFER_DATA (buf), num_samples);
+
+ return GST_FLOW_OK;
+}