/* * 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; }