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gstreamer/gst/audiofx/audioecho.c
Sebastian Dröge 1f32369451 Limit the delay by a new max-delay property 
Introduce a new max-delay property that can only
be set before going to PLAYING or PAUSED. This
is used to limit the maximum delay and is set
to the current delay by default.

Using this will make sure that we have enough data
in our internal ringbuffer for the echo. With dynamic
reallocation of the ringbuffer as used before silence
could've been used as the echo directly after setting
a new delay.
2009-01-28 16:01:34 +01:00

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/*
* 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: 0.10.12
*
* 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:{
guint 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:{
guint 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;
}
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