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Diffstat (limited to 'gst/audiofx/audioecho.c')
| -rw-r--r-- | gst/audiofx/audioecho.c | 367 |
1 files changed, 367 insertions, 0 deletions
diff --git a/gst/audiofx/audioecho.c b/gst/audiofx/audioecho.c new file mode 100644 index 00000000..04d51240 --- /dev/null +++ b/gst/audiofx/audioecho.c @@ -0,0 +1,367 @@ +/* + * 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 + * + * <refsect2> + * audioecho adds an echo or reverb effect to an audio stream. The echo + * delay, intensity and the percentage of feedback can be configured. + * <para> + * <programlisting> + * 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 + * </programlisting> + * </para> + * </refsect2> + * + * Since: 0.10.12 + */ + +#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_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 nanosecondsecho", 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->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 rate, width, channels; + + GST_BASE_TRANSFORM_LOCK (self); + self->delay = g_value_get_uint64 (value); + + rate = GST_AUDIO_FILTER (self)->format.rate; + width = GST_AUDIO_FILTER (self)->format.width / 8; + channels = GST_AUDIO_FILTER (self)->format.channels; + + if (self->buffer && rate > 0) { + guint new_echo = + MAX (gst_util_uint64_scale (self->delay, rate, GST_SECOND), 1); + guint new_size = new_echo * width * channels; + + if (new_size > self->buffer_size) { + guint i; + guint8 *old_buffer = self->buffer; + + self->buffer_size = new_size; + self->buffer = g_malloc0 (new_size); + + for (i = 0; i < self->buffer_size_frames; i++) { + memcpy (&self->buffer[i * width * channels], + &old_buffer[((i + + self->buffer_pos) % self->buffer_size_frames) * + width * channels], channels * width); + } + self->buffer_size_frames = self->delay_frames = new_echo; + self->buffer_pos = 0; + } + } else if (self->buffer) { + g_free (self->buffer); + self->buffer = NULL; + } + + 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_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 (self->delay_frames, 1000); + self->buffer_size = self->buffer_size_frames * width * channels; + self->buffer = g_malloc0 (self->buffer_size); + self->buffer_pos = 0; + } + + self->process (self, GST_BUFFER_DATA (buf), num_samples); + + return GST_FLOW_OK; +} |