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., 51 Franklin St, Fifth Floor,
18 * Boston, MA 02110-1301, USA.
22 * SECTION:element-audioecho
24 * audioecho adds an echo or (simple) reverb effect to an audio stream. The echo
25 * delay, intensity and the percentage of feedback can be configured.
27 * For getting an echo effect you have to set the delay to a larger value,
28 * for example 200ms and more. Everything below will result in a simple
29 * reverb effect, which results in a slightly metallic sound.
31 * Use the max-delay property to set the maximum amount of delay that
32 * will be used. This can only be set before going to the PAUSED or PLAYING
33 * state and will be set to the current delay by default.
36 * <title>Example launch line</title>
38 * gst-launch-1.0 autoaudiosrc ! audioconvert ! audioecho delay=500000000 intensity=0.6 feedback=0.4 ! audioconvert ! autoaudiosink
39 * gst-launch-1.0 filesrc location="melo1.ogg" ! decodebin ! audioconvert ! audioecho delay=50000000 intensity=0.6 feedback=0.4 ! audioconvert ! autoaudiosink
49 #include <gst/base/gstbasetransform.h>
50 #include <gst/audio/audio.h>
51 #include <gst/audio/gstaudiofilter.h>
53 #include "audioecho.h"
55 #define GST_CAT_DEFAULT gst_audio_echo_debug
56 GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
67 #define ALLOWED_CAPS \
69 " format=(string) {"GST_AUDIO_NE(F32)","GST_AUDIO_NE(F64)"}, " \
70 " rate=(int)[1,MAX]," \
71 " channels=(int)[1,MAX]," \
72 " layout=(string) interleaved"
74 #define gst_audio_echo_parent_class parent_class
75 G_DEFINE_TYPE (GstAudioEcho, gst_audio_echo, GST_TYPE_AUDIO_FILTER);
77 static void gst_audio_echo_set_property (GObject * object, guint prop_id,
78 const GValue * value, GParamSpec * pspec);
79 static void gst_audio_echo_get_property (GObject * object, guint prop_id,
80 GValue * value, GParamSpec * pspec);
81 static void gst_audio_echo_finalize (GObject * object);
83 static gboolean gst_audio_echo_setup (GstAudioFilter * self,
84 const GstAudioInfo * info);
85 static gboolean gst_audio_echo_stop (GstBaseTransform * base);
86 static GstFlowReturn gst_audio_echo_transform_ip (GstBaseTransform * base,
89 static void gst_audio_echo_transform_float (GstAudioEcho * self,
90 gfloat * data, guint num_samples);
91 static void gst_audio_echo_transform_double (GstAudioEcho * self,
92 gdouble * data, guint num_samples);
94 /* GObject vmethod implementations */
97 gst_audio_echo_class_init (GstAudioEchoClass * klass)
99 GObjectClass *gobject_class = (GObjectClass *) klass;
100 GstElementClass *gstelement_class = (GstElementClass *) klass;
101 GstBaseTransformClass *basetransform_class = (GstBaseTransformClass *) klass;
102 GstAudioFilterClass *audioself_class = (GstAudioFilterClass *) klass;
105 GST_DEBUG_CATEGORY_INIT (gst_audio_echo_debug, "audioecho", 0,
106 "audioecho element");
108 gobject_class->set_property = gst_audio_echo_set_property;
109 gobject_class->get_property = gst_audio_echo_get_property;
110 gobject_class->finalize = gst_audio_echo_finalize;
112 g_object_class_install_property (gobject_class, PROP_DELAY,
113 g_param_spec_uint64 ("delay", "Delay",
114 "Delay of the echo in nanoseconds", 1, G_MAXUINT64,
115 1, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS
116 | GST_PARAM_CONTROLLABLE));
118 g_object_class_install_property (gobject_class, PROP_MAX_DELAY,
119 g_param_spec_uint64 ("max-delay", "Maximum Delay",
120 "Maximum delay of the echo in nanoseconds"
121 " (can't be changed in PLAYING or PAUSED state)",
123 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS |
124 GST_PARAM_MUTABLE_READY));
126 g_object_class_install_property (gobject_class, PROP_INTENSITY,
127 g_param_spec_float ("intensity", "Intensity",
128 "Intensity of the echo", 0.0, 1.0,
129 0.0, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS
130 | GST_PARAM_CONTROLLABLE));
132 g_object_class_install_property (gobject_class, PROP_FEEDBACK,
133 g_param_spec_float ("feedback", "Feedback",
134 "Amount of feedback", 0.0, 1.0,
135 0.0, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS
136 | GST_PARAM_CONTROLLABLE));
138 gst_element_class_set_static_metadata (gstelement_class, "Audio echo",
139 "Filter/Effect/Audio",
140 "Adds an echo or reverb effect to an audio stream",
141 "Sebastian Dröge <sebastian.droege@collabora.co.uk>");
143 caps = gst_caps_from_string (ALLOWED_CAPS);
144 gst_audio_filter_class_add_pad_templates (GST_AUDIO_FILTER_CLASS (klass),
146 gst_caps_unref (caps);
148 audioself_class->setup = GST_DEBUG_FUNCPTR (gst_audio_echo_setup);
149 basetransform_class->transform_ip =
150 GST_DEBUG_FUNCPTR (gst_audio_echo_transform_ip);
151 basetransform_class->stop = GST_DEBUG_FUNCPTR (gst_audio_echo_stop);
155 gst_audio_echo_init (GstAudioEcho * self)
159 self->intensity = 0.0;
160 self->feedback = 0.0;
162 g_mutex_init (&self->lock);
164 gst_base_transform_set_in_place (GST_BASE_TRANSFORM (self), TRUE);
168 gst_audio_echo_finalize (GObject * object)
170 GstAudioEcho *self = GST_AUDIO_ECHO (object);
172 g_free (self->buffer);
175 g_mutex_clear (&self->lock);
177 G_OBJECT_CLASS (parent_class)->finalize (object);
181 gst_audio_echo_set_property (GObject * object, guint prop_id,
182 const GValue * value, GParamSpec * pspec)
184 GstAudioEcho *self = GST_AUDIO_ECHO (object);
188 guint64 max_delay, delay;
191 g_mutex_lock (&self->lock);
192 delay = g_value_get_uint64 (value);
193 max_delay = self->max_delay;
195 if (delay > max_delay && GST_STATE (self) > GST_STATE_READY) {
196 GST_WARNING_OBJECT (self, "New delay (%" GST_TIME_FORMAT ") "
197 "is larger than maximum delay (%" GST_TIME_FORMAT ")",
198 GST_TIME_ARGS (delay), GST_TIME_ARGS (max_delay));
199 self->delay = max_delay;
202 self->max_delay = MAX (delay, max_delay);
204 rate = GST_AUDIO_FILTER_RATE (self);
207 MAX (gst_util_uint64_scale (self->delay, rate, GST_SECOND), 1);
209 g_mutex_unlock (&self->lock);
212 case PROP_MAX_DELAY:{
213 guint64 max_delay, delay;
215 g_mutex_lock (&self->lock);
216 max_delay = g_value_get_uint64 (value);
219 if (GST_STATE (self) > GST_STATE_READY) {
220 GST_ERROR_OBJECT (self, "Can't change maximum delay in"
221 " PLAYING or PAUSED state");
224 self->max_delay = max_delay;
226 g_mutex_unlock (&self->lock);
229 case PROP_INTENSITY:{
230 g_mutex_lock (&self->lock);
231 self->intensity = g_value_get_float (value);
232 g_mutex_unlock (&self->lock);
236 g_mutex_lock (&self->lock);
237 self->feedback = g_value_get_float (value);
238 g_mutex_unlock (&self->lock);
242 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
248 gst_audio_echo_get_property (GObject * object, guint prop_id,
249 GValue * value, GParamSpec * pspec)
251 GstAudioEcho *self = GST_AUDIO_ECHO (object);
255 g_mutex_lock (&self->lock);
256 g_value_set_uint64 (value, self->delay);
257 g_mutex_unlock (&self->lock);
260 g_mutex_lock (&self->lock);
261 g_value_set_uint64 (value, self->max_delay);
262 g_mutex_unlock (&self->lock);
265 g_mutex_lock (&self->lock);
266 g_value_set_float (value, self->intensity);
267 g_mutex_unlock (&self->lock);
270 g_mutex_lock (&self->lock);
271 g_value_set_float (value, self->feedback);
272 g_mutex_unlock (&self->lock);
275 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
280 /* GstAudioFilter vmethod implementations */
283 gst_audio_echo_setup (GstAudioFilter * base, const GstAudioInfo * info)
285 GstAudioEcho *self = GST_AUDIO_ECHO (base);
288 switch (GST_AUDIO_INFO_FORMAT (info)) {
289 case GST_AUDIO_FORMAT_F32:
290 self->process = (GstAudioEchoProcessFunc)
291 gst_audio_echo_transform_float;
293 case GST_AUDIO_FORMAT_F64:
294 self->process = (GstAudioEchoProcessFunc)
295 gst_audio_echo_transform_double;
302 g_free (self->buffer);
304 self->buffer_pos = 0;
305 self->buffer_size = 0;
306 self->buffer_size_frames = 0;
312 gst_audio_echo_stop (GstBaseTransform * base)
314 GstAudioEcho *self = GST_AUDIO_ECHO (base);
316 g_free (self->buffer);
318 self->buffer_pos = 0;
319 self->buffer_size = 0;
320 self->buffer_size_frames = 0;
325 #define TRANSFORM_FUNC(name, type) \
327 gst_audio_echo_transform_##name (GstAudioEcho * self, \
328 type * data, guint num_samples) \
330 type *buffer = (type *) self->buffer; \
331 guint channels = GST_AUDIO_FILTER_CHANNELS (self); \
332 guint rate = GST_AUDIO_FILTER_RATE (self); \
334 guint echo_index = self->buffer_size_frames - self->delay_frames; \
335 gdouble echo_off = ((((gdouble) self->delay) * rate) / GST_SECOND) - self->delay_frames; \
337 if (echo_off < 0.0) \
340 num_samples /= channels; \
342 for (i = 0; i < num_samples; i++) { \
343 guint echo0_index = ((echo_index + self->buffer_pos) % self->buffer_size_frames) * channels; \
344 guint echo1_index = ((echo_index + self->buffer_pos +1) % self->buffer_size_frames) * channels; \
345 guint rbout_index = (self->buffer_pos % self->buffer_size_frames) * channels; \
346 for (j = 0; j < channels; j++) { \
347 gdouble in = data[i*channels + j]; \
348 gdouble echo0 = buffer[echo0_index + j]; \
349 gdouble echo1 = buffer[echo1_index + j]; \
350 gdouble echo = echo0 + (echo1-echo0)*echo_off; \
351 type out = in + self->intensity * echo; \
353 data[i*channels + j] = out; \
355 buffer[rbout_index + j] = in + self->feedback * echo; \
357 self->buffer_pos = (self->buffer_pos + 1) % self->buffer_size_frames; \
361 TRANSFORM_FUNC (float, gfloat);
362 TRANSFORM_FUNC (double, gdouble);
364 /* GstBaseTransform vmethod implementations */
366 gst_audio_echo_transform_ip (GstBaseTransform * base, GstBuffer * buf)
368 GstAudioEcho *self = GST_AUDIO_ECHO (base);
370 GstClockTime timestamp, stream_time;
373 g_mutex_lock (&self->lock);
374 timestamp = GST_BUFFER_TIMESTAMP (buf);
376 gst_segment_to_stream_time (&base->segment, GST_FORMAT_TIME, timestamp);
378 GST_DEBUG_OBJECT (self, "sync to %" GST_TIME_FORMAT,
379 GST_TIME_ARGS (timestamp));
381 if (GST_CLOCK_TIME_IS_VALID (stream_time))
382 gst_object_sync_values (GST_OBJECT (self), stream_time);
384 if (self->buffer == NULL) {
387 bpf = GST_AUDIO_FILTER_BPF (self);
388 rate = GST_AUDIO_FILTER_RATE (self);
391 MAX (gst_util_uint64_scale (self->delay, rate, GST_SECOND), 1);
392 self->buffer_size_frames =
393 MAX (gst_util_uint64_scale (self->max_delay, rate, GST_SECOND), 1);
395 self->buffer_size = self->buffer_size_frames * bpf;
396 self->buffer = g_try_malloc0 (self->buffer_size);
397 self->buffer_pos = 0;
399 if (self->buffer == NULL) {
400 g_mutex_unlock (&self->lock);
401 GST_ERROR_OBJECT (self, "Failed to allocate %u bytes", self->buffer_size);
402 return GST_FLOW_ERROR;
406 gst_buffer_map (buf, &map, GST_MAP_READWRITE);
407 num_samples = map.size / GST_AUDIO_FILTER_BPS (self);
409 self->process (self, map.data, num_samples);
411 gst_buffer_unmap (buf, &map);
412 g_mutex_unlock (&self->lock);