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gstreamer/gst/audiofx/audioecho.c
Tim-Philipp Müller 230cf41cc9 Fix FSF address 
https://bugzilla.gnome.org/show_bug.cgi?id=687520
2012-11-04 00:07:18 +00:00

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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., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
/**
* SECTION:element-audioecho
* @Since: 0.10.14
*
* 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-1.0 filesrc location="melo1.ogg" ! audioconvert ! audioecho delay=500000000 intensity=0.6 feedback=0.4 ! audioconvert ! autoaudiosink
* gst-launch-1.0 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 "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," \
" format=(string) {"GST_AUDIO_NE(F32)","GST_AUDIO_NE(F64)"}, " \
" rate=(int)[1,MAX]," \
" channels=(int)[1,MAX]," \
" layout=(string) interleaved"
#define gst_audio_echo_parent_class parent_class
G_DEFINE_TYPE (GstAudioEcho, gst_audio_echo, GST_TYPE_AUDIO_FILTER);
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,
const GstAudioInfo * info);
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_class_init (GstAudioEchoClass * klass)
{
GObjectClass *gobject_class = (GObjectClass *) klass;
GstElementClass *gstelement_class = (GstElementClass *) klass;
GstBaseTransformClass *basetransform_class = (GstBaseTransformClass *) klass;
GstAudioFilterClass *audioself_class = (GstAudioFilterClass *) klass;
GstCaps *caps;
GST_DEBUG_CATEGORY_INIT (gst_audio_echo_debug, "audioecho", 0,
"audioecho element");
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_MUTABLE_READY));
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));
gst_element_class_set_static_metadata (gstelement_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);
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)
{
self->delay = 1;
self->max_delay = 1;
self->intensity = 0.0;
self->feedback = 0.0;
g_mutex_init (&self->lock);
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_mutex_clear (&self->lock);
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:{
guint64 max_delay, delay;
g_mutex_lock (&self->lock);
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);
}
g_mutex_unlock (&self->lock);
break;
}
case PROP_MAX_DELAY:{
guint64 max_delay, delay;
g_mutex_lock (&self->lock);
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;
}
g_mutex_unlock (&self->lock);
break;
}
case PROP_INTENSITY:{
g_mutex_lock (&self->lock);
self->intensity = g_value_get_float (value);
g_mutex_unlock (&self->lock);
break;
}
case PROP_FEEDBACK:{
g_mutex_lock (&self->lock);
self->feedback = g_value_get_float (value);
g_mutex_unlock (&self->lock);
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:
g_mutex_lock (&self->lock);
g_value_set_uint64 (value, self->delay);
g_mutex_unlock (&self->lock);
break;
case PROP_MAX_DELAY:
g_mutex_lock (&self->lock);
g_value_set_uint64 (value, self->max_delay);
g_mutex_unlock (&self->lock);
break;
case PROP_INTENSITY:
g_mutex_lock (&self->lock);
g_value_set_float (value, self->intensity);
g_mutex_unlock (&self->lock);
break;
case PROP_FEEDBACK:
g_mutex_lock (&self->lock);
g_value_set_float (value, self->feedback);
g_mutex_unlock (&self->lock);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
/* GstAudioFilter vmethod implementations */
static gboolean
gst_audio_echo_setup (GstAudioFilter * base, const GstAudioInfo * info)
{
GstAudioEcho *self = GST_AUDIO_ECHO (base);
gboolean ret = TRUE;
switch (GST_AUDIO_INFO_FORMAT (info)) {
case GST_AUDIO_FORMAT_F32:
self->process = (GstAudioEchoProcessFunc)
gst_audio_echo_transform_float;
break;
case GST_AUDIO_FORMAT_F64:
self->process = (GstAudioEchoProcessFunc)
gst_audio_echo_transform_double;
break;
default:
ret = FALSE;
break;
}
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_CHANNELS (self); \
guint rate = GST_AUDIO_FILTER_RATE (self); \
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;
GstClockTime timestamp, stream_time;
GstMapInfo map;
g_mutex_lock (&self->lock);
timestamp = GST_BUFFER_TIMESTAMP (buf);
stream_time =
gst_segment_to_stream_time (&base->segment, GST_FORMAT_TIME, timestamp);
GST_DEBUG_OBJECT (self, "sync to %" GST_TIME_FORMAT,
GST_TIME_ARGS (timestamp));
if (GST_CLOCK_TIME_IS_VALID (stream_time))
gst_object_sync_values (GST_OBJECT (self), stream_time);
if (self->buffer == NULL) {
guint bpf, rate;
bpf = GST_AUDIO_FILTER_BPF (self);
rate = GST_AUDIO_FILTER_RATE (self);
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 * bpf;
self->buffer = g_try_malloc0 (self->buffer_size);
self->buffer_pos = 0;
if (self->buffer == NULL) {
g_mutex_unlock (&self->lock);
GST_ERROR_OBJECT (self, "Failed to allocate %u bytes", self->buffer_size);
return GST_FLOW_ERROR;
}
}
gst_buffer_map (buf, &map, GST_MAP_READWRITE);
num_samples = map.size / GST_AUDIO_FILTER_BPS (self);
self->process (self, map.data, num_samples);
gst_buffer_unmap (buf, &map);
g_mutex_unlock (&self->lock);
return GST_FLOW_OK;
}
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