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504 lines
16 KiB
C
504 lines
16 KiB
C
/*
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* GStreamer
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* Copyright (C) 2009 Sebastian Dröge <sebastian.droege@collabora.co.uk>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, USA.
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*/
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/**
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* SECTION:element-audioecho
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* @title: audioecho
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*
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* audioecho adds an echo or (simple) reverb effect to an audio stream. The echo
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* delay, intensity and the percentage of feedback can be configured.
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*
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* For getting an echo effect you have to set the delay to a larger value,
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* for example 200ms and more. Everything below will result in a simple
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* reverb effect, which results in a slightly metallic sound.
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*
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* Use the max-delay property to set the maximum amount of delay that
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* will be used. This can only be set before going to the PAUSED or PLAYING
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* state and will be set to the current delay by default.
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*
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* audioecho can also be used to apply a configurable delay to audio channels
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* by setting surround-delay=true. In that mode, it just delays "surround
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* channels" by the delay amount instead of performing an echo. The
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* channels that are configured surround channels for the delay are
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* selected using the surround-channels mask property.
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*
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* ## Example launch lines
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* |[
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* gst-launch-1.0 autoaudiosrc ! audioconvert ! audioecho delay=500000000 intensity=0.6 feedback=0.4 ! audioconvert ! autoaudiosink
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* gst-launch-1.0 filesrc location="melo1.ogg" ! decodebin ! audioconvert ! audioecho delay=50000000 intensity=0.6 feedback=0.4 ! audioconvert ! autoaudiosink
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* gst-launch-1.0 audiotestsrc ! audioconvert ! audio/x-raw,channels=4 ! audioecho surround-delay=true delay=500000000 ! audioconvert ! autoaudiosink
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* ]|
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*
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*/
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#ifdef HAVE_CONFIG_H
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#include "config.h"
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#endif
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#include <gst/gst.h>
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#include <gst/base/gstbasetransform.h>
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#include <gst/audio/audio.h>
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#include <gst/audio/gstaudiofilter.h>
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#include "audioecho.h"
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#define GST_CAT_DEFAULT gst_audio_echo_debug
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GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
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/* Everything except the first 2 channels are considered surround */
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#define DEFAULT_SURROUND_MASK ~((guint64)(0x3))
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enum
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{
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PROP_0,
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PROP_DELAY,
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PROP_MAX_DELAY,
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PROP_INTENSITY,
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PROP_FEEDBACK,
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PROP_SUR_DELAY,
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PROP_SUR_MASK
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};
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#define ALLOWED_CAPS \
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"audio/x-raw," \
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" format=(string) {"GST_AUDIO_NE(F32)","GST_AUDIO_NE(F64)"}, " \
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" rate=(int)[1,MAX]," \
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" channels=(int)[1,MAX]," \
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" layout=(string) interleaved"
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#define gst_audio_echo_parent_class parent_class
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G_DEFINE_TYPE (GstAudioEcho, gst_audio_echo, GST_TYPE_AUDIO_FILTER);
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static void gst_audio_echo_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec);
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static void gst_audio_echo_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec);
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static void gst_audio_echo_finalize (GObject * object);
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static gboolean gst_audio_echo_setup (GstAudioFilter * self,
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const GstAudioInfo * info);
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static gboolean gst_audio_echo_stop (GstBaseTransform * base);
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static GstFlowReturn gst_audio_echo_transform_ip (GstBaseTransform * base,
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GstBuffer * buf);
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static void gst_audio_echo_transform_float (GstAudioEcho * self,
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gfloat * data, guint num_samples);
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static void gst_audio_echo_transform_double (GstAudioEcho * self,
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gdouble * data, guint num_samples);
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/* GObject vmethod implementations */
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static void
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gst_audio_echo_class_init (GstAudioEchoClass * klass)
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{
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GObjectClass *gobject_class = (GObjectClass *) klass;
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GstElementClass *gstelement_class = (GstElementClass *) klass;
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GstBaseTransformClass *basetransform_class = (GstBaseTransformClass *) klass;
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GstAudioFilterClass *audioself_class = (GstAudioFilterClass *) klass;
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GstCaps *caps;
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GST_DEBUG_CATEGORY_INIT (gst_audio_echo_debug, "audioecho", 0,
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"audioecho element");
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gobject_class->set_property = gst_audio_echo_set_property;
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gobject_class->get_property = gst_audio_echo_get_property;
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gobject_class->finalize = gst_audio_echo_finalize;
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g_object_class_install_property (gobject_class, PROP_DELAY,
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g_param_spec_uint64 ("delay", "Delay",
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"Delay of the echo in nanoseconds", 1, G_MAXUINT64,
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1, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS
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| GST_PARAM_CONTROLLABLE));
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g_object_class_install_property (gobject_class, PROP_MAX_DELAY,
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g_param_spec_uint64 ("max-delay", "Maximum Delay",
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"Maximum delay of the echo in nanoseconds"
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" (can't be changed in PLAYING or PAUSED state)",
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1, G_MAXUINT64, 1,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS |
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GST_PARAM_MUTABLE_READY));
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g_object_class_install_property (gobject_class, PROP_INTENSITY,
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g_param_spec_float ("intensity", "Intensity",
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"Intensity of the echo", 0.0, 1.0,
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0.0, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS
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| GST_PARAM_CONTROLLABLE));
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g_object_class_install_property (gobject_class, PROP_FEEDBACK,
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g_param_spec_float ("feedback", "Feedback",
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"Amount of feedback", 0.0, 1.0,
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0.0, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS
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| GST_PARAM_CONTROLLABLE));
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g_object_class_install_property (gobject_class, PROP_SUR_DELAY,
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g_param_spec_boolean ("surround-delay", "Enable Surround Delay",
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"Delay Surround Channels when TRUE instead of applying an echo effect",
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FALSE,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS | GST_PARAM_CONTROLLABLE));
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g_object_class_install_property (gobject_class, PROP_SUR_MASK,
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g_param_spec_uint64 ("surround-mask", "Surround Mask",
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"A bitmask of channels that are considered surround and delayed when surround-delay = TRUE",
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1, G_MAXUINT64, DEFAULT_SURROUND_MASK,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS |
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GST_PARAM_MUTABLE_READY));
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gst_element_class_set_static_metadata (gstelement_class, "Audio echo",
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"Filter/Effect/Audio",
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"Adds an echo or reverb effect to an audio stream",
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"Sebastian Dröge <sebastian.droege@collabora.co.uk>");
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caps = gst_caps_from_string (ALLOWED_CAPS);
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gst_audio_filter_class_add_pad_templates (GST_AUDIO_FILTER_CLASS (klass),
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caps);
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gst_caps_unref (caps);
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audioself_class->setup = GST_DEBUG_FUNCPTR (gst_audio_echo_setup);
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basetransform_class->transform_ip =
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GST_DEBUG_FUNCPTR (gst_audio_echo_transform_ip);
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basetransform_class->stop = GST_DEBUG_FUNCPTR (gst_audio_echo_stop);
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}
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static void
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gst_audio_echo_init (GstAudioEcho * self)
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{
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self->delay = 1;
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self->max_delay = 1;
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self->intensity = 0.0;
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self->feedback = 0.0;
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self->surdelay = FALSE;
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self->surround_mask = DEFAULT_SURROUND_MASK;
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g_mutex_init (&self->lock);
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gst_base_transform_set_in_place (GST_BASE_TRANSFORM (self), TRUE);
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}
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static void
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gst_audio_echo_finalize (GObject * object)
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{
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GstAudioEcho *self = GST_AUDIO_ECHO (object);
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g_free (self->buffer);
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self->buffer = NULL;
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g_mutex_clear (&self->lock);
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G_OBJECT_CLASS (parent_class)->finalize (object);
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}
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static void
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gst_audio_echo_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec)
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{
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GstAudioEcho *self = GST_AUDIO_ECHO (object);
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switch (prop_id) {
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case PROP_DELAY:{
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guint64 max_delay, delay;
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guint rate;
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g_mutex_lock (&self->lock);
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delay = g_value_get_uint64 (value);
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max_delay = self->max_delay;
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if (delay > max_delay && GST_STATE (self) > GST_STATE_READY) {
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GST_WARNING_OBJECT (self, "New delay (%" GST_TIME_FORMAT ") "
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"is larger than maximum delay (%" GST_TIME_FORMAT ")",
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GST_TIME_ARGS (delay), GST_TIME_ARGS (max_delay));
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self->delay = max_delay;
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} else {
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self->delay = delay;
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self->max_delay = MAX (delay, max_delay);
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if (delay > max_delay) {
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g_free (self->buffer);
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self->buffer = NULL;
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}
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}
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rate = GST_AUDIO_FILTER_RATE (self);
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if (rate > 0)
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self->delay_frames =
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MAX (gst_util_uint64_scale (self->delay, rate, GST_SECOND), 1);
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g_mutex_unlock (&self->lock);
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break;
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}
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case PROP_MAX_DELAY:{
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guint64 max_delay;
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g_mutex_lock (&self->lock);
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max_delay = g_value_get_uint64 (value);
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if (GST_STATE (self) > GST_STATE_READY) {
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GST_ERROR_OBJECT (self, "Can't change maximum delay in"
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" PLAYING or PAUSED state");
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} else {
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self->max_delay = max_delay;
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g_free (self->buffer);
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self->buffer = NULL;
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}
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g_mutex_unlock (&self->lock);
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break;
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}
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case PROP_INTENSITY:{
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g_mutex_lock (&self->lock);
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self->intensity = g_value_get_float (value);
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g_mutex_unlock (&self->lock);
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break;
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}
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case PROP_FEEDBACK:{
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g_mutex_lock (&self->lock);
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self->feedback = g_value_get_float (value);
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g_mutex_unlock (&self->lock);
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break;
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}
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case PROP_SUR_DELAY:{
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g_mutex_lock (&self->lock);
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self->surdelay = g_value_get_boolean (value);
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g_mutex_unlock (&self->lock);
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break;
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}
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case PROP_SUR_MASK:{
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g_mutex_lock (&self->lock);
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self->surround_mask = g_value_get_uint64 (value);
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g_mutex_unlock (&self->lock);
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break;
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}
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
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break;
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}
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}
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static void
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gst_audio_echo_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec)
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{
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GstAudioEcho *self = GST_AUDIO_ECHO (object);
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switch (prop_id) {
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case PROP_DELAY:
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g_mutex_lock (&self->lock);
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g_value_set_uint64 (value, self->delay);
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g_mutex_unlock (&self->lock);
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break;
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case PROP_MAX_DELAY:
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g_mutex_lock (&self->lock);
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g_value_set_uint64 (value, self->max_delay);
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g_mutex_unlock (&self->lock);
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break;
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case PROP_INTENSITY:
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g_mutex_lock (&self->lock);
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g_value_set_float (value, self->intensity);
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g_mutex_unlock (&self->lock);
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break;
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case PROP_FEEDBACK:
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g_mutex_lock (&self->lock);
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g_value_set_float (value, self->feedback);
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g_mutex_unlock (&self->lock);
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break;
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case PROP_SUR_DELAY:
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g_mutex_lock (&self->lock);
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g_value_set_boolean (value, self->surdelay);
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g_mutex_unlock (&self->lock);
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break;
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case PROP_SUR_MASK:{
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g_mutex_lock (&self->lock);
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g_value_set_uint64 (value, self->surround_mask);
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g_mutex_unlock (&self->lock);
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break;
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}
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
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break;
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}
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}
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/* GstAudioFilter vmethod implementations */
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static gboolean
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gst_audio_echo_setup (GstAudioFilter * base, const GstAudioInfo * info)
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{
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GstAudioEcho *self = GST_AUDIO_ECHO (base);
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gboolean ret = TRUE;
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switch (GST_AUDIO_INFO_FORMAT (info)) {
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case GST_AUDIO_FORMAT_F32:
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self->process = (GstAudioEchoProcessFunc)
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gst_audio_echo_transform_float;
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break;
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case GST_AUDIO_FORMAT_F64:
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self->process = (GstAudioEchoProcessFunc)
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gst_audio_echo_transform_double;
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break;
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default:
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ret = FALSE;
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break;
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}
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g_free (self->buffer);
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self->buffer = NULL;
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self->buffer_pos = 0;
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self->buffer_size = 0;
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self->buffer_size_frames = 0;
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return ret;
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}
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static gboolean
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gst_audio_echo_stop (GstBaseTransform * base)
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{
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GstAudioEcho *self = GST_AUDIO_ECHO (base);
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g_free (self->buffer);
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self->buffer = NULL;
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self->buffer_pos = 0;
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self->buffer_size = 0;
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self->buffer_size_frames = 0;
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return TRUE;
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}
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#define TRANSFORM_FUNC(name, type) \
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static void \
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gst_audio_echo_transform_##name (GstAudioEcho * self, \
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type * data, guint num_samples) \
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{ \
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type *buffer = (type *) self->buffer; \
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guint channels = GST_AUDIO_FILTER_CHANNELS (self); \
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guint i, j; \
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guint echo_offset = self->buffer_size_frames - self->delay_frames; \
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gdouble intensity = self->intensity; \
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gdouble feedback = self->feedback; \
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guint buffer_pos = self->buffer_pos; \
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guint buffer_size_frames = self->buffer_size_frames; \
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\
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if (self->surdelay == FALSE) { \
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guint read_pos = ((echo_offset + buffer_pos) % buffer_size_frames) * channels; \
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guint write_pos = (buffer_pos % buffer_size_frames) * channels; \
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guint buffer_size = buffer_size_frames * channels; \
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for (i = 0; i < num_samples; i++) { \
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gdouble in = *data; \
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gdouble echo = buffer[read_pos]; \
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type out = in + intensity * echo; \
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\
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*data = out; \
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\
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buffer[write_pos] = in + feedback * echo; \
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read_pos = (read_pos + 1) % buffer_size; \
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write_pos = (write_pos + 1) % buffer_size; \
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data++; \
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} \
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buffer_pos = write_pos / channels; \
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} else { \
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guint64 surround_mask = self->surround_mask; \
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guint read_pos = ((echo_offset + buffer_pos) % buffer_size_frames) * channels; \
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guint write_pos = (buffer_pos % buffer_size_frames) * channels; \
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guint buffer_size = buffer_size_frames * channels; \
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\
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num_samples /= channels; \
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\
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for (i = 0; i < num_samples; i++) { \
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guint64 channel_mask = 1; \
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\
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for (j = 0; j < channels; j++) { \
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if (channel_mask & surround_mask) { \
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gdouble in = data[j]; \
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gdouble echo = buffer[read_pos + j]; \
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type out = echo; \
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\
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data[j] = out; \
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\
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buffer[write_pos + j] = in; \
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} else { \
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gdouble in = data[j]; \
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gdouble echo = buffer[read_pos + j]; \
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type out = in + intensity * echo; \
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\
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data[j] = out; \
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\
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buffer[write_pos + j] = in + feedback * echo; \
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} \
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channel_mask <<= 1; \
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} \
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read_pos = (read_pos + channels) % buffer_size; \
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write_pos = (write_pos + channels) % buffer_size; \
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data += channels; \
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} \
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buffer_pos = write_pos / channels; \
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} \
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self->buffer_pos = buffer_pos; \
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}
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TRANSFORM_FUNC (float, gfloat);
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TRANSFORM_FUNC (double, gdouble);
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/* GstBaseTransform vmethod implementations */
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static GstFlowReturn
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gst_audio_echo_transform_ip (GstBaseTransform * base, GstBuffer * buf)
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{
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GstAudioEcho *self = GST_AUDIO_ECHO (base);
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guint num_samples;
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GstClockTime timestamp, stream_time;
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GstMapInfo map;
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g_mutex_lock (&self->lock);
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timestamp = GST_BUFFER_TIMESTAMP (buf);
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stream_time =
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gst_segment_to_stream_time (&base->segment, GST_FORMAT_TIME, timestamp);
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GST_DEBUG_OBJECT (self, "sync to %" GST_TIME_FORMAT,
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GST_TIME_ARGS (timestamp));
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if (GST_CLOCK_TIME_IS_VALID (stream_time))
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gst_object_sync_values (GST_OBJECT (self), stream_time);
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if (self->buffer == NULL) {
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guint bpf, rate;
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|
|
|
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;
|
|
}
|