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 | GST_PARAM_CONTROLLABLE));
142 g_object_class_install_property (gobject_class, PROP_INTENSITY,
143 g_param_spec_float ("intensity", "Intensity",
144 "Intensity of the echo", 0.0, 1.0,
145 0.0, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS
146 | GST_PARAM_CONTROLLABLE));
148 g_object_class_install_property (gobject_class, PROP_FEEDBACK,
149 g_param_spec_float ("feedback", "Feedback",
150 "Amount of feedback", 0.0, 1.0,
151 0.0, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS
152 | GST_PARAM_CONTROLLABLE));
154 audioself_class->setup = GST_DEBUG_FUNCPTR (gst_audio_echo_setup);
155 basetransform_class->transform_ip =
156 GST_DEBUG_FUNCPTR (gst_audio_echo_transform_ip);
157 basetransform_class->stop = GST_DEBUG_FUNCPTR (gst_audio_echo_stop);
161 gst_audio_echo_init (GstAudioEcho * self, GstAudioEchoClass * klass)
165 self->intensity = 0.0;
166 self->feedback = 0.0;
168 gst_base_transform_set_in_place (GST_BASE_TRANSFORM (self), TRUE);
172 gst_audio_echo_finalize (GObject * object)
174 GstAudioEcho *self = GST_AUDIO_ECHO (object);
176 g_free (self->buffer);
179 G_OBJECT_CLASS (parent_class)->finalize (object);
183 gst_audio_echo_set_property (GObject * object, guint prop_id,
184 const GValue * value, GParamSpec * pspec)
186 GstAudioEcho *self = GST_AUDIO_ECHO (object);
190 guint64 max_delay, delay;
192 GST_BASE_TRANSFORM_LOCK (self);
193 delay = g_value_get_uint64 (value);
194 max_delay = self->max_delay;
196 if (delay > max_delay && GST_STATE (self) > GST_STATE_READY) {
197 GST_WARNING_OBJECT (self, "New delay (%" GST_TIME_FORMAT ") "
198 "is larger than maximum delay (%" GST_TIME_FORMAT ")",
199 GST_TIME_ARGS (delay), GST_TIME_ARGS (max_delay));
200 self->delay = max_delay;
203 self->max_delay = MAX (delay, max_delay);
205 GST_BASE_TRANSFORM_UNLOCK (self);
208 case PROP_MAX_DELAY:{
209 guint64 max_delay, delay;
211 GST_BASE_TRANSFORM_LOCK (self);
212 max_delay = g_value_get_uint64 (value);
215 if (GST_STATE (self) > GST_STATE_READY) {
216 GST_ERROR_OBJECT (self, "Can't change maximum delay in"
217 " PLAYING or PAUSED state");
220 self->max_delay = max_delay;
222 GST_BASE_TRANSFORM_UNLOCK (self);
225 case PROP_INTENSITY:{
226 GST_BASE_TRANSFORM_LOCK (self);
227 self->intensity = g_value_get_float (value);
228 GST_BASE_TRANSFORM_UNLOCK (self);
232 GST_BASE_TRANSFORM_LOCK (self);
233 self->feedback = g_value_get_float (value);
234 GST_BASE_TRANSFORM_UNLOCK (self);
238 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
244 gst_audio_echo_get_property (GObject * object, guint prop_id,
245 GValue * value, GParamSpec * pspec)
247 GstAudioEcho *self = GST_AUDIO_ECHO (object);
251 GST_BASE_TRANSFORM_LOCK (self);
252 g_value_set_uint64 (value, self->delay);
253 GST_BASE_TRANSFORM_UNLOCK (self);
256 GST_BASE_TRANSFORM_LOCK (self);
257 g_value_set_uint64 (value, self->max_delay);
258 GST_BASE_TRANSFORM_UNLOCK (self);
261 GST_BASE_TRANSFORM_LOCK (self);
262 g_value_set_float (value, self->intensity);
263 GST_BASE_TRANSFORM_UNLOCK (self);
266 GST_BASE_TRANSFORM_LOCK (self);
267 g_value_set_float (value, self->feedback);
268 GST_BASE_TRANSFORM_UNLOCK (self);
271 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
276 /* GstAudioFilter vmethod implementations */
279 gst_audio_echo_setup (GstAudioFilter * base, GstRingBufferSpec * format)
281 GstAudioEcho *self = GST_AUDIO_ECHO (base);
284 if (format->type == GST_BUFTYPE_FLOAT && format->width == 32)
285 self->process = (GstAudioEchoProcessFunc)
286 gst_audio_echo_transform_float;
287 else if (format->type == GST_BUFTYPE_FLOAT && format->width == 64)
288 self->process = (GstAudioEchoProcessFunc)
289 gst_audio_echo_transform_double;
293 g_free (self->buffer);
295 self->buffer_pos = 0;
296 self->buffer_size = 0;
297 self->buffer_size_frames = 0;
303 gst_audio_echo_stop (GstBaseTransform * base)
305 GstAudioEcho *self = GST_AUDIO_ECHO (base);
307 g_free (self->buffer);
309 self->buffer_pos = 0;
310 self->buffer_size = 0;
311 self->buffer_size_frames = 0;
316 #define TRANSFORM_FUNC(name, type) \
318 gst_audio_echo_transform_##name (GstAudioEcho * self, \
319 type * data, guint num_samples) \
321 type *buffer = (type *) self->buffer; \
322 guint channels = GST_AUDIO_FILTER (self)->format.channels; \
323 guint rate = GST_AUDIO_FILTER (self)->format.rate; \
325 guint echo_index = self->buffer_size_frames - self->delay_frames; \
326 gdouble echo_off = ((((gdouble) self->delay) * rate) / GST_SECOND) - self->delay_frames; \
328 if (echo_off < 0.0) \
331 num_samples /= channels; \
333 for (i = 0; i < num_samples; i++) { \
334 guint echo0_index = ((echo_index + self->buffer_pos) % self->buffer_size_frames) * channels; \
335 guint echo1_index = ((echo_index + self->buffer_pos +1) % self->buffer_size_frames) * channels; \
336 guint rbout_index = (self->buffer_pos % self->buffer_size_frames) * channels; \
337 for (j = 0; j < channels; j++) { \
338 gdouble in = data[i*channels + j]; \
339 gdouble echo0 = buffer[echo0_index + j]; \
340 gdouble echo1 = buffer[echo1_index + j]; \
341 gdouble echo = echo0 + (echo1-echo0)*echo_off; \
342 type out = in + self->intensity * echo; \
344 data[i*channels + j] = out; \
346 buffer[rbout_index + j] = in + self->feedback * echo; \
348 self->buffer_pos = (self->buffer_pos + 1) % self->buffer_size_frames; \
352 TRANSFORM_FUNC (float, gfloat);
353 TRANSFORM_FUNC (double, gdouble);
355 /* GstBaseTransform vmethod implementations */
357 gst_audio_echo_transform_ip (GstBaseTransform * base, GstBuffer * buf)
359 GstAudioEcho *self = GST_AUDIO_ECHO (base);
361 GstClockTime timestamp, stream_time;
363 timestamp = GST_BUFFER_TIMESTAMP (buf);
365 gst_segment_to_stream_time (&base->segment, GST_FORMAT_TIME, timestamp);
367 GST_DEBUG_OBJECT (self, "sync to %" GST_TIME_FORMAT,
368 GST_TIME_ARGS (timestamp));
370 if (GST_CLOCK_TIME_IS_VALID (stream_time))
371 gst_object_sync_values (G_OBJECT (self), stream_time);
374 GST_BUFFER_SIZE (buf) / (GST_AUDIO_FILTER (self)->format.width / 8);
376 if (self->buffer == NULL) {
377 guint width, rate, channels;
379 width = GST_AUDIO_FILTER (self)->format.width / 8;
380 rate = GST_AUDIO_FILTER (self)->format.rate;
381 channels = GST_AUDIO_FILTER (self)->format.channels;
384 MAX (gst_util_uint64_scale (self->delay, rate, GST_SECOND), 1);
385 self->buffer_size_frames =
386 MAX (gst_util_uint64_scale (self->max_delay, rate, GST_SECOND), 1);
388 self->buffer_size = self->buffer_size_frames * width * channels;
389 self->buffer = g_try_malloc0 (self->buffer_size);
390 self->buffer_pos = 0;
392 if (self->buffer == NULL) {
393 GST_ERROR_OBJECT (self, "Failed to allocate %u bytes", self->buffer_size);
394 return GST_FLOW_ERROR;
398 self->process (self, GST_BUFFER_DATA (buf), num_samples);