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
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-1.0 filesrc location="melo1.ogg" ! audioconvert ! audioecho delay=500000000 intensity=0.6 feedback=0.4 ! audioconvert ! autoaudiosink
40 * gst-launch-1.0 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>
54 #include "audioecho.h"
56 #define GST_CAT_DEFAULT gst_audio_echo_debug
57 GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
68 #define ALLOWED_CAPS \
70 " format=(string) {"GST_AUDIO_NE(F32)","GST_AUDIO_NE(F64)"}, " \
71 " rate=(int)[1,MAX]," \
72 " channels=(int)[1,MAX]," \
73 " layout=(string) interleaved"
75 #define gst_audio_echo_parent_class parent_class
76 G_DEFINE_TYPE (GstAudioEcho, gst_audio_echo, GST_TYPE_AUDIO_FILTER);
78 static void gst_audio_echo_set_property (GObject * object, guint prop_id,
79 const GValue * value, GParamSpec * pspec);
80 static void gst_audio_echo_get_property (GObject * object, guint prop_id,
81 GValue * value, GParamSpec * pspec);
82 static void gst_audio_echo_finalize (GObject * object);
84 static gboolean gst_audio_echo_setup (GstAudioFilter * self,
85 const GstAudioInfo * info);
86 static gboolean gst_audio_echo_stop (GstBaseTransform * base);
87 static GstFlowReturn gst_audio_echo_transform_ip (GstBaseTransform * base,
90 static void gst_audio_echo_transform_float (GstAudioEcho * self,
91 gfloat * data, guint num_samples);
92 static void gst_audio_echo_transform_double (GstAudioEcho * self,
93 gdouble * data, guint num_samples);
95 /* GObject vmethod implementations */
98 gst_audio_echo_class_init (GstAudioEchoClass * klass)
100 GObjectClass *gobject_class = (GObjectClass *) klass;
101 GstElementClass *gstelement_class = (GstElementClass *) klass;
102 GstBaseTransformClass *basetransform_class = (GstBaseTransformClass *) klass;
103 GstAudioFilterClass *audioself_class = (GstAudioFilterClass *) klass;
106 GST_DEBUG_CATEGORY_INIT (gst_audio_echo_debug, "audioecho", 0,
107 "audioecho element");
109 gobject_class->set_property = gst_audio_echo_set_property;
110 gobject_class->get_property = gst_audio_echo_get_property;
111 gobject_class->finalize = gst_audio_echo_finalize;
113 g_object_class_install_property (gobject_class, PROP_DELAY,
114 g_param_spec_uint64 ("delay", "Delay",
115 "Delay of the echo in nanoseconds", 1, G_MAXUINT64,
116 1, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS
117 | GST_PARAM_CONTROLLABLE));
119 g_object_class_install_property (gobject_class, PROP_MAX_DELAY,
120 g_param_spec_uint64 ("max-delay", "Maximum Delay",
121 "Maximum delay of the echo in nanoseconds"
122 " (can't be changed in PLAYING or PAUSED state)",
124 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS |
125 GST_PARAM_MUTABLE_READY));
127 g_object_class_install_property (gobject_class, PROP_INTENSITY,
128 g_param_spec_float ("intensity", "Intensity",
129 "Intensity of the echo", 0.0, 1.0,
130 0.0, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS
131 | GST_PARAM_CONTROLLABLE));
133 g_object_class_install_property (gobject_class, PROP_FEEDBACK,
134 g_param_spec_float ("feedback", "Feedback",
135 "Amount of feedback", 0.0, 1.0,
136 0.0, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS
137 | GST_PARAM_CONTROLLABLE));
139 gst_element_class_set_static_metadata (gstelement_class, "Audio echo",
140 "Filter/Effect/Audio",
141 "Adds an echo or reverb effect to an audio stream",
142 "Sebastian Dröge <sebastian.droege@collabora.co.uk>");
144 caps = gst_caps_from_string (ALLOWED_CAPS);
145 gst_audio_filter_class_add_pad_templates (GST_AUDIO_FILTER_CLASS (klass),
147 gst_caps_unref (caps);
149 audioself_class->setup = GST_DEBUG_FUNCPTR (gst_audio_echo_setup);
150 basetransform_class->transform_ip =
151 GST_DEBUG_FUNCPTR (gst_audio_echo_transform_ip);
152 basetransform_class->stop = GST_DEBUG_FUNCPTR (gst_audio_echo_stop);
156 gst_audio_echo_init (GstAudioEcho * self)
160 self->intensity = 0.0;
161 self->feedback = 0.0;
163 g_mutex_init (&self->lock);
165 gst_base_transform_set_in_place (GST_BASE_TRANSFORM (self), TRUE);
169 gst_audio_echo_finalize (GObject * object)
171 GstAudioEcho *self = GST_AUDIO_ECHO (object);
173 g_free (self->buffer);
176 g_mutex_clear (&self->lock);
178 G_OBJECT_CLASS (parent_class)->finalize (object);
182 gst_audio_echo_set_property (GObject * object, guint prop_id,
183 const GValue * value, GParamSpec * pspec)
185 GstAudioEcho *self = GST_AUDIO_ECHO (object);
189 guint64 max_delay, delay;
192 g_mutex_lock (&self->lock);
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 rate = GST_AUDIO_FILTER_RATE (self);
208 MAX (gst_util_uint64_scale (self->delay, rate, GST_SECOND), 1);
210 g_mutex_unlock (&self->lock);
213 case PROP_MAX_DELAY:{
214 guint64 max_delay, delay;
216 g_mutex_lock (&self->lock);
217 max_delay = g_value_get_uint64 (value);
220 if (GST_STATE (self) > GST_STATE_READY) {
221 GST_ERROR_OBJECT (self, "Can't change maximum delay in"
222 " PLAYING or PAUSED state");
225 self->max_delay = max_delay;
227 g_mutex_unlock (&self->lock);
230 case PROP_INTENSITY:{
231 g_mutex_lock (&self->lock);
232 self->intensity = g_value_get_float (value);
233 g_mutex_unlock (&self->lock);
237 g_mutex_lock (&self->lock);
238 self->feedback = g_value_get_float (value);
239 g_mutex_unlock (&self->lock);
243 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
249 gst_audio_echo_get_property (GObject * object, guint prop_id,
250 GValue * value, GParamSpec * pspec)
252 GstAudioEcho *self = GST_AUDIO_ECHO (object);
256 g_mutex_lock (&self->lock);
257 g_value_set_uint64 (value, self->delay);
258 g_mutex_unlock (&self->lock);
261 g_mutex_lock (&self->lock);
262 g_value_set_uint64 (value, self->max_delay);
263 g_mutex_unlock (&self->lock);
266 g_mutex_lock (&self->lock);
267 g_value_set_float (value, self->intensity);
268 g_mutex_unlock (&self->lock);
271 g_mutex_lock (&self->lock);
272 g_value_set_float (value, self->feedback);
273 g_mutex_unlock (&self->lock);
276 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
281 /* GstAudioFilter vmethod implementations */
284 gst_audio_echo_setup (GstAudioFilter * base, const GstAudioInfo * info)
286 GstAudioEcho *self = GST_AUDIO_ECHO (base);
289 switch (GST_AUDIO_INFO_FORMAT (info)) {
290 case GST_AUDIO_FORMAT_F32:
291 self->process = (GstAudioEchoProcessFunc)
292 gst_audio_echo_transform_float;
294 case GST_AUDIO_FORMAT_F64:
295 self->process = (GstAudioEchoProcessFunc)
296 gst_audio_echo_transform_double;
303 g_free (self->buffer);
305 self->buffer_pos = 0;
306 self->buffer_size = 0;
307 self->buffer_size_frames = 0;
313 gst_audio_echo_stop (GstBaseTransform * base)
315 GstAudioEcho *self = GST_AUDIO_ECHO (base);
317 g_free (self->buffer);
319 self->buffer_pos = 0;
320 self->buffer_size = 0;
321 self->buffer_size_frames = 0;
326 #define TRANSFORM_FUNC(name, type) \
328 gst_audio_echo_transform_##name (GstAudioEcho * self, \
329 type * data, guint num_samples) \
331 type *buffer = (type *) self->buffer; \
332 guint channels = GST_AUDIO_FILTER_CHANNELS (self); \
333 guint rate = GST_AUDIO_FILTER_RATE (self); \
335 guint echo_index = self->buffer_size_frames - self->delay_frames; \
336 gdouble echo_off = ((((gdouble) self->delay) * rate) / GST_SECOND) - self->delay_frames; \
338 if (echo_off < 0.0) \
341 num_samples /= channels; \
343 for (i = 0; i < num_samples; i++) { \
344 guint echo0_index = ((echo_index + self->buffer_pos) % self->buffer_size_frames) * channels; \
345 guint echo1_index = ((echo_index + self->buffer_pos +1) % self->buffer_size_frames) * channels; \
346 guint rbout_index = (self->buffer_pos % self->buffer_size_frames) * channels; \
347 for (j = 0; j < channels; j++) { \
348 gdouble in = data[i*channels + j]; \
349 gdouble echo0 = buffer[echo0_index + j]; \
350 gdouble echo1 = buffer[echo1_index + j]; \
351 gdouble echo = echo0 + (echo1-echo0)*echo_off; \
352 type out = in + self->intensity * echo; \
354 data[i*channels + j] = out; \
356 buffer[rbout_index + j] = in + self->feedback * echo; \
358 self->buffer_pos = (self->buffer_pos + 1) % self->buffer_size_frames; \
362 TRANSFORM_FUNC (float, gfloat);
363 TRANSFORM_FUNC (double, gdouble);
365 /* GstBaseTransform vmethod implementations */
367 gst_audio_echo_transform_ip (GstBaseTransform * base, GstBuffer * buf)
369 GstAudioEcho *self = GST_AUDIO_ECHO (base);
371 GstClockTime timestamp, stream_time;
374 g_mutex_lock (&self->lock);
375 timestamp = GST_BUFFER_TIMESTAMP (buf);
377 gst_segment_to_stream_time (&base->segment, GST_FORMAT_TIME, timestamp);
379 GST_DEBUG_OBJECT (self, "sync to %" GST_TIME_FORMAT,
380 GST_TIME_ARGS (timestamp));
382 if (GST_CLOCK_TIME_IS_VALID (stream_time))
383 gst_object_sync_values (GST_OBJECT (self), stream_time);
385 if (self->buffer == NULL) {
388 bpf = GST_AUDIO_FILTER_BPF (self);
389 rate = GST_AUDIO_FILTER_RATE (self);
392 MAX (gst_util_uint64_scale (self->delay, rate, GST_SECOND), 1);
393 self->buffer_size_frames =
394 MAX (gst_util_uint64_scale (self->max_delay, rate, GST_SECOND), 1);
396 self->buffer_size = self->buffer_size_frames * bpf;
397 self->buffer = g_try_malloc0 (self->buffer_size);
398 self->buffer_pos = 0;
400 if (self->buffer == NULL) {
401 g_mutex_unlock (&self->lock);
402 GST_ERROR_OBJECT (self, "Failed to allocate %u bytes", self->buffer_size);
403 return GST_FLOW_ERROR;
407 gst_buffer_map (buf, &map, GST_MAP_READWRITE);
408 num_samples = map.size / GST_AUDIO_FILTER_BPS (self);
410 self->process (self, map.data, num_samples);
412 gst_buffer_unmap (buf, &map);
413 g_mutex_unlock (&self->lock);