3 * Copyright (C) 2009 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5 * This library is free software; you can redistribute it and/or
6 * modify it under the terms of the GNU Library General Public
7 * License as published by the Free Software Foundation; either
8 * version 2 of the License, or (at your option) any later version.
10 * This library is distributed in the hope that it will be useful,
11 * but WITHOUT ANY WARRANTY; without even the implied warranty of
12 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
13 * Library General Public License for more details.
15 * You should have received a copy of the GNU Library General Public
16 * License along with this library; if not, write to the
17 * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
18 * Boston, MA 02111-1307, USA.
22 * SECTION:element-audioecho
25 * audioecho adds an echo or (simple) reverb effect to an audio stream. The echo
26 * delay, intensity and the percentage of feedback can be configured.
28 * For getting an echo effect you have to set the delay to a larger value,
29 * for example 200ms and more. Everything below will result in a simple
30 * reverb effect, which results in a slightly metallic sound.
32 * Use the max-delay property to set the maximum amount of delay that
33 * will be used. This can only be set before going to the PAUSED or PLAYING
34 * state and will be set to the current delay by default.
37 * <title>Example launch line</title>
39 * gst-launch filesrc location="melo1.ogg" ! audioconvert ! audioecho delay=500000000 intensity=0.6 feedback=0.4 ! audioconvert ! autoaudiosink
40 * gst-launch filesrc location="melo1.ogg" ! decodebin ! audioconvert ! audioecho delay=50000000 intensity=0.6 feedback=0.4 ! audioconvert ! autoaudiosink
50 #include <gst/base/gstbasetransform.h>
51 #include <gst/audio/audio.h>
52 #include <gst/audio/gstaudiofilter.h>
53 #include <gst/controller/gstcontroller.h>
55 #include "audioecho.h"
57 #define GST_CAT_DEFAULT gst_audio_echo_debug
58 GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
69 #define ALLOWED_CAPS \
70 "audio/x-raw-float," \
71 " width=(int) { 32, 64 }, " \
72 " endianness=(int)BYTE_ORDER," \
73 " rate=(int)[1,MAX]," \
74 " channels=(int)[1,MAX]"
76 #define DEBUG_INIT(bla) \
77 GST_DEBUG_CATEGORY_INIT (gst_audio_echo_debug, "audioecho", 0, "audioecho element");
79 GST_BOILERPLATE_FULL (GstAudioEcho, gst_audio_echo, GstAudioFilter,
80 GST_TYPE_AUDIO_FILTER, DEBUG_INIT);
82 static void gst_audio_echo_set_property (GObject * object, guint prop_id,
83 const GValue * value, GParamSpec * pspec);
84 static void gst_audio_echo_get_property (GObject * object, guint prop_id,
85 GValue * value, GParamSpec * pspec);
86 static void gst_audio_echo_finalize (GObject * object);
88 static gboolean gst_audio_echo_setup (GstAudioFilter * self,
89 GstRingBufferSpec * format);
90 static gboolean gst_audio_echo_stop (GstBaseTransform * base);
91 static GstFlowReturn gst_audio_echo_transform_ip (GstBaseTransform * base,
94 static void gst_audio_echo_transform_float (GstAudioEcho * self,
95 gfloat * data, guint num_samples);
96 static void gst_audio_echo_transform_double (GstAudioEcho * self,
97 gdouble * data, guint num_samples);
99 /* GObject vmethod implementations */
102 gst_audio_echo_base_init (gpointer klass)
104 GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
107 gst_element_class_set_details_simple (element_class, "Audio echo",
108 "Filter/Effect/Audio",
109 "Adds an echo or reverb effect to an audio stream",
110 "Sebastian Dröge <sebastian.droege@collabora.co.uk>");
112 caps = gst_caps_from_string (ALLOWED_CAPS);
113 gst_audio_filter_class_add_pad_templates (GST_AUDIO_FILTER_CLASS (klass),
115 gst_caps_unref (caps);
119 gst_audio_echo_class_init (GstAudioEchoClass * klass)
121 GObjectClass *gobject_class = (GObjectClass *) klass;
122 GstBaseTransformClass *basetransform_class = (GstBaseTransformClass *) klass;
123 GstAudioFilterClass *audioself_class = (GstAudioFilterClass *) klass;
125 gobject_class->set_property = gst_audio_echo_set_property;
126 gobject_class->get_property = gst_audio_echo_get_property;
127 gobject_class->finalize = gst_audio_echo_finalize;
129 g_object_class_install_property (gobject_class, PROP_DELAY,
130 g_param_spec_uint64 ("delay", "Delay",
131 "Delay of the echo in nanoseconds", 1, G_MAXUINT64,
132 1, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS
133 | GST_PARAM_CONTROLLABLE));
135 g_object_class_install_property (gobject_class, PROP_MAX_DELAY,
136 g_param_spec_uint64 ("max-delay", "Maximum Delay",
137 "Maximum delay of the echo in nanoseconds"
138 " (can't be changed in PLAYING or PAUSED state)",
140 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS |
141 GST_PARAM_MUTABLE_READY));
143 g_object_class_install_property (gobject_class, PROP_INTENSITY,
144 g_param_spec_float ("intensity", "Intensity",
145 "Intensity of the echo", 0.0, 1.0,
146 0.0, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS
147 | GST_PARAM_CONTROLLABLE));
149 g_object_class_install_property (gobject_class, PROP_FEEDBACK,
150 g_param_spec_float ("feedback", "Feedback",
151 "Amount of feedback", 0.0, 1.0,
152 0.0, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS
153 | GST_PARAM_CONTROLLABLE));
155 audioself_class->setup = GST_DEBUG_FUNCPTR (gst_audio_echo_setup);
156 basetransform_class->transform_ip =
157 GST_DEBUG_FUNCPTR (gst_audio_echo_transform_ip);
158 basetransform_class->stop = GST_DEBUG_FUNCPTR (gst_audio_echo_stop);
162 gst_audio_echo_init (GstAudioEcho * self, GstAudioEchoClass * klass)
166 self->intensity = 0.0;
167 self->feedback = 0.0;
169 gst_base_transform_set_in_place (GST_BASE_TRANSFORM (self), TRUE);
173 gst_audio_echo_finalize (GObject * object)
175 GstAudioEcho *self = GST_AUDIO_ECHO (object);
177 g_free (self->buffer);
180 G_OBJECT_CLASS (parent_class)->finalize (object);
184 gst_audio_echo_set_property (GObject * object, guint prop_id,
185 const GValue * value, GParamSpec * pspec)
187 GstAudioEcho *self = GST_AUDIO_ECHO (object);
191 guint64 max_delay, delay;
193 GST_BASE_TRANSFORM_LOCK (self);
194 delay = g_value_get_uint64 (value);
195 max_delay = self->max_delay;
197 if (delay > max_delay && GST_STATE (self) > GST_STATE_READY) {
198 GST_WARNING_OBJECT (self, "New delay (%" GST_TIME_FORMAT ") "
199 "is larger than maximum delay (%" GST_TIME_FORMAT ")",
200 GST_TIME_ARGS (delay), GST_TIME_ARGS (max_delay));
201 self->delay = max_delay;
204 self->max_delay = MAX (delay, max_delay);
206 GST_BASE_TRANSFORM_UNLOCK (self);
209 case PROP_MAX_DELAY:{
210 guint64 max_delay, delay;
212 GST_BASE_TRANSFORM_LOCK (self);
213 max_delay = g_value_get_uint64 (value);
216 if (GST_STATE (self) > GST_STATE_READY) {
217 GST_ERROR_OBJECT (self, "Can't change maximum delay in"
218 " PLAYING or PAUSED state");
221 self->max_delay = max_delay;
223 GST_BASE_TRANSFORM_UNLOCK (self);
226 case PROP_INTENSITY:{
227 GST_BASE_TRANSFORM_LOCK (self);
228 self->intensity = g_value_get_float (value);
229 GST_BASE_TRANSFORM_UNLOCK (self);
233 GST_BASE_TRANSFORM_LOCK (self);
234 self->feedback = g_value_get_float (value);
235 GST_BASE_TRANSFORM_UNLOCK (self);
239 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
245 gst_audio_echo_get_property (GObject * object, guint prop_id,
246 GValue * value, GParamSpec * pspec)
248 GstAudioEcho *self = GST_AUDIO_ECHO (object);
252 GST_BASE_TRANSFORM_LOCK (self);
253 g_value_set_uint64 (value, self->delay);
254 GST_BASE_TRANSFORM_UNLOCK (self);
257 GST_BASE_TRANSFORM_LOCK (self);
258 g_value_set_uint64 (value, self->max_delay);
259 GST_BASE_TRANSFORM_UNLOCK (self);
262 GST_BASE_TRANSFORM_LOCK (self);
263 g_value_set_float (value, self->intensity);
264 GST_BASE_TRANSFORM_UNLOCK (self);
267 GST_BASE_TRANSFORM_LOCK (self);
268 g_value_set_float (value, self->feedback);
269 GST_BASE_TRANSFORM_UNLOCK (self);
272 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
277 /* GstAudioFilter vmethod implementations */
280 gst_audio_echo_setup (GstAudioFilter * base, GstRingBufferSpec * format)
282 GstAudioEcho *self = GST_AUDIO_ECHO (base);
285 if (format->type == GST_BUFTYPE_FLOAT && format->width == 32)
286 self->process = (GstAudioEchoProcessFunc)
287 gst_audio_echo_transform_float;
288 else if (format->type == GST_BUFTYPE_FLOAT && format->width == 64)
289 self->process = (GstAudioEchoProcessFunc)
290 gst_audio_echo_transform_double;
294 g_free (self->buffer);
296 self->buffer_pos = 0;
297 self->buffer_size = 0;
298 self->buffer_size_frames = 0;
304 gst_audio_echo_stop (GstBaseTransform * base)
306 GstAudioEcho *self = GST_AUDIO_ECHO (base);
308 g_free (self->buffer);
310 self->buffer_pos = 0;
311 self->buffer_size = 0;
312 self->buffer_size_frames = 0;
317 #define TRANSFORM_FUNC(name, type) \
319 gst_audio_echo_transform_##name (GstAudioEcho * self, \
320 type * data, guint num_samples) \
322 type *buffer = (type *) self->buffer; \
323 guint channels = GST_AUDIO_FILTER (self)->format.channels; \
324 guint rate = GST_AUDIO_FILTER (self)->format.rate; \
326 guint echo_index = self->buffer_size_frames - self->delay_frames; \
327 gdouble echo_off = ((((gdouble) self->delay) * rate) / GST_SECOND) - self->delay_frames; \
329 if (echo_off < 0.0) \
332 num_samples /= channels; \
334 for (i = 0; i < num_samples; i++) { \
335 guint echo0_index = ((echo_index + self->buffer_pos) % self->buffer_size_frames) * channels; \
336 guint echo1_index = ((echo_index + self->buffer_pos +1) % self->buffer_size_frames) * channels; \
337 guint rbout_index = (self->buffer_pos % self->buffer_size_frames) * channels; \
338 for (j = 0; j < channels; j++) { \
339 gdouble in = data[i*channels + j]; \
340 gdouble echo0 = buffer[echo0_index + j]; \
341 gdouble echo1 = buffer[echo1_index + j]; \
342 gdouble echo = echo0 + (echo1-echo0)*echo_off; \
343 type out = in + self->intensity * echo; \
345 data[i*channels + j] = out; \
347 buffer[rbout_index + j] = in + self->feedback * echo; \
349 self->buffer_pos = (self->buffer_pos + 1) % self->buffer_size_frames; \
353 TRANSFORM_FUNC (float, gfloat);
354 TRANSFORM_FUNC (double, gdouble);
356 /* GstBaseTransform vmethod implementations */
358 gst_audio_echo_transform_ip (GstBaseTransform * base, GstBuffer * buf)
360 GstAudioEcho *self = GST_AUDIO_ECHO (base);
362 GstClockTime timestamp, stream_time;
364 timestamp = GST_BUFFER_TIMESTAMP (buf);
366 gst_segment_to_stream_time (&base->segment, GST_FORMAT_TIME, timestamp);
368 GST_DEBUG_OBJECT (self, "sync to %" GST_TIME_FORMAT,
369 GST_TIME_ARGS (timestamp));
371 if (GST_CLOCK_TIME_IS_VALID (stream_time))
372 gst_object_sync_values (G_OBJECT (self), stream_time);
375 GST_BUFFER_SIZE (buf) / (GST_AUDIO_FILTER (self)->format.width / 8);
377 if (self->buffer == NULL) {
378 guint width, rate, channels;
380 width = GST_AUDIO_FILTER (self)->format.width / 8;
381 rate = GST_AUDIO_FILTER (self)->format.rate;
382 channels = GST_AUDIO_FILTER (self)->format.channels;
385 MAX (gst_util_uint64_scale (self->delay, rate, GST_SECOND), 1);
386 self->buffer_size_frames =
387 MAX (gst_util_uint64_scale (self->max_delay, rate, GST_SECOND), 1);
389 self->buffer_size = self->buffer_size_frames * width * channels;
390 self->buffer = g_try_malloc0 (self->buffer_size);
391 self->buffer_pos = 0;
393 if (self->buffer == NULL) {
394 GST_ERROR_OBJECT (self, "Failed to allocate %u bytes", self->buffer_size);
395 return GST_FLOW_ERROR;
399 self->process (self, GST_BUFFER_DATA (buf), num_samples);