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f13c8b6576
Original commit message from CVS: * gst/audiofx/audioamplify.c: * gst/audiofx/audiochebband.c: * gst/audiofx/audiocheblimit.c: * gst/audiofx/audiodynamic.c: * gst/audiofx/audioinvert.c: * gst/audiofx/audiopanorama.c: * gst/audiofx/audiowsincband.c: * gst/audiofx/audiowsinclimit.c: Fix long description of audiofx elements. Fixes bug #515457.
800 lines
25 KiB
C
800 lines
25 KiB
C
/* -*- c-basic-offset: 2 -*-
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*
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* GStreamer
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* Copyright (C) 1999-2001 Erik Walthinsen <omega@cse.ogi.edu>
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* 2006 Dreamlab Technologies Ltd. <mathis.hofer@dreamlab.net>
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* 2007 Sebastian Dröge <slomo@circular-chaos.org>
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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., 59 Temple Place - Suite 330,
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* Boston, MA 02111-1307, USA.
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*
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*
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* this windowed sinc filter is taken from the freely downloadable DSP book,
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* "The Scientist and Engineer's Guide to Digital Signal Processing",
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* chapter 16
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* available at http://www.dspguide.com/
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*
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* TODO: - Implement the convolution in place, probably only makes sense
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* when using FFT convolution as currently the convolution itself
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* is probably the bottleneck
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* - Maybe allow cascading the filter to get a better stopband attenuation.
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* Can be done by convolving a filter kernel with itself
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*/
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/**
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* SECTION:element-audiowsinclimit
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* @short_description: Windowed Sinc low pass and high pass filter
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*
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* <refsect2>
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* <para>
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* Attenuates all frequencies above the cutoff frequency (low-pass) or all frequencies below the
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* cutoff frequency (high-pass). The length parameter controls the rolloff, the window parameter
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* controls rolloff and stopband attenuation. The Hamming window provides a faster rolloff but a bit
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* worse stopband attenuation, the other way around for the Blackman window.
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* </para>
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* <para>
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* This element has the advantage over the Chebyshev lowpass and highpass filter that it has
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* a much better rolloff when using a larger kernel size and almost linear phase. The only
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* disadvantage is the much slower execution time with larger kernels.
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* </para>
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* <title>Example launch line</title>
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* <para>
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* <programlisting>
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* gst-launch audiotestsrc freq=1500 ! audioconvert ! audiowsinclimit mode=low-pass frequency=1000 length=501 ! audioconvert ! alsasink
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* gst-launch filesrc location="melo1.ogg" ! oggdemux ! vorbisdec ! audioconvert ! audiowsinclimit mode=high-pass frequency=15000 length=501 ! audioconvert ! alsasink
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* gst-launch audiotestsrc wave=white-noise ! audioconvert ! audiowsinclimit mode=low-pass frequency=1000 length=10001 window=blackman ! audioconvert ! alsasink
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* </programlisting>
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* </para>
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* </refsect2>
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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 <string.h>
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#include <math.h>
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#include <gst/gst.h>
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#include <gst/audio/gstaudiofilter.h>
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#include <gst/controller/gstcontroller.h>
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#include "audiowsinclimit.h"
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#define GST_CAT_DEFAULT audio_wsinclimit_debug
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GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
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static const GstElementDetails audio_wsinclimit_details =
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GST_ELEMENT_DETAILS ("Low pass & high pass filter",
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"Filter/Effect/Audio",
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"Low pass and high pass windowed sinc filter",
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"Thomas <thomas@apestaart.org>, "
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"Steven W. Smith, "
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"Dreamlab Technologies Ltd. <mathis.hofer@dreamlab.net>, "
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"Sebastian Dröge <slomo@circular-chaos.org>");
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/* Filter signals and args */
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enum
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{
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/* FILL ME */
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LAST_SIGNAL
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};
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enum
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{
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PROP_0,
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PROP_LENGTH,
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PROP_FREQUENCY,
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PROP_MODE,
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PROP_WINDOW
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};
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enum
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{
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MODE_LOW_PASS = 0,
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MODE_HIGH_PASS
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};
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#define GST_TYPE_AUDIO_WSINC_LIMIT_MODE (audio_wsinclimit_mode_get_type ())
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static GType
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audio_wsinclimit_mode_get_type (void)
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{
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static GType gtype = 0;
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if (gtype == 0) {
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static const GEnumValue values[] = {
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{MODE_LOW_PASS, "Low pass (default)",
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"low-pass"},
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{MODE_HIGH_PASS, "High pass",
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"high-pass"},
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{0, NULL, NULL}
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};
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gtype = g_enum_register_static ("GstAudioWSincLimitMode", values);
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}
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return gtype;
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}
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enum
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{
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WINDOW_HAMMING = 0,
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WINDOW_BLACKMAN
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};
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#define GST_TYPE_AUDIO_WSINC_LIMIT_WINDOW (audio_wsinclimit_window_get_type ())
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static GType
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audio_wsinclimit_window_get_type (void)
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{
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static GType gtype = 0;
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if (gtype == 0) {
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static const GEnumValue values[] = {
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{WINDOW_HAMMING, "Hamming window (default)",
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"hamming"},
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{WINDOW_BLACKMAN, "Blackman window",
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"blackman"},
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{0, NULL, NULL}
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};
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gtype = g_enum_register_static ("GstAudioWSincLimitWindow", values);
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}
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return gtype;
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}
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#define ALLOWED_CAPS \
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"audio/x-raw-float, " \
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" width = (int) { 32, 64 }, " \
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" endianness = (int) BYTE_ORDER, " \
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" rate = (int) [ 1, MAX ], " \
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" channels = (int) [ 1, MAX ]"
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#define DEBUG_INIT(bla) \
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GST_DEBUG_CATEGORY_INIT (audio_wsinclimit_debug, "audiowsinclimit", 0, \
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"Low-pass and High-pass Windowed sinc filter plugin");
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GST_BOILERPLATE_FULL (GstAudioWSincLimit, audio_wsinclimit, GstAudioFilter,
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GST_TYPE_AUDIO_FILTER, DEBUG_INIT);
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static void audio_wsinclimit_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec);
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static void audio_wsinclimit_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec);
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static GstFlowReturn audio_wsinclimit_transform (GstBaseTransform * base,
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GstBuffer * inbuf, GstBuffer * outbuf);
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static gboolean audio_wsinclimit_start (GstBaseTransform * base);
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static gboolean audio_wsinclimit_event (GstBaseTransform * base,
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GstEvent * event);
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static gboolean audio_wsinclimit_setup (GstAudioFilter * base,
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GstRingBufferSpec * format);
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static gboolean audio_wsinclimit_query (GstPad * pad, GstQuery * query);
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static const GstQueryType *audio_wsinclimit_query_type (GstPad * pad);
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/* Element class */
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static void
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audio_wsinclimit_dispose (GObject * object)
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{
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GstAudioWSincLimit *self = GST_AUDIO_WSINC_LIMIT (object);
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if (self->residue) {
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g_free (self->residue);
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self->residue = NULL;
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}
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if (self->kernel) {
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g_free (self->kernel);
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self->kernel = NULL;
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}
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G_OBJECT_CLASS (parent_class)->dispose (object);
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}
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static void
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audio_wsinclimit_base_init (gpointer g_class)
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{
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GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
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GstCaps *caps;
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gst_element_class_set_details (element_class, &audio_wsinclimit_details);
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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 (g_class),
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caps);
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gst_caps_unref (caps);
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}
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static void
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audio_wsinclimit_class_init (GstAudioWSincLimitClass * klass)
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{
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GObjectClass *gobject_class;
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GstBaseTransformClass *trans_class;
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GstAudioFilterClass *filter_class;
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gobject_class = (GObjectClass *) klass;
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trans_class = (GstBaseTransformClass *) klass;
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filter_class = (GstAudioFilterClass *) klass;
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gobject_class->set_property = audio_wsinclimit_set_property;
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gobject_class->get_property = audio_wsinclimit_get_property;
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gobject_class->dispose = audio_wsinclimit_dispose;
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/* FIXME: Don't use the complete possible range but restrict the upper boundary
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* so automatically generated UIs can use a slider */
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g_object_class_install_property (gobject_class, PROP_FREQUENCY,
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g_param_spec_float ("cutoff", "Cutoff",
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"Cut-off Frequency (Hz)", 0.0, 100000.0, 0.0,
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G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
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g_object_class_install_property (gobject_class, PROP_LENGTH,
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g_param_spec_int ("length", "Length",
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"Filter kernel length, will be rounded to the next odd number",
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3, 50000, 101, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
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g_object_class_install_property (gobject_class, PROP_MODE,
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g_param_spec_enum ("mode", "Mode",
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"Low pass or high pass mode", GST_TYPE_AUDIO_WSINC_LIMIT_MODE,
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MODE_LOW_PASS, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
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g_object_class_install_property (gobject_class, PROP_WINDOW,
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g_param_spec_enum ("window", "Window",
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"Window function to use", GST_TYPE_AUDIO_WSINC_LIMIT_WINDOW,
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WINDOW_HAMMING, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
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trans_class->transform = GST_DEBUG_FUNCPTR (audio_wsinclimit_transform);
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trans_class->start = GST_DEBUG_FUNCPTR (audio_wsinclimit_start);
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trans_class->event = GST_DEBUG_FUNCPTR (audio_wsinclimit_event);
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filter_class->setup = GST_DEBUG_FUNCPTR (audio_wsinclimit_setup);
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}
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static void
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audio_wsinclimit_init (GstAudioWSincLimit * self,
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GstAudioWSincLimitClass * g_class)
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{
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self->mode = MODE_LOW_PASS;
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self->window = WINDOW_HAMMING;
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self->kernel_length = 101;
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self->latency = 50;
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self->cutoff = 0.0;
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self->kernel = NULL;
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self->residue = NULL;
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self->have_kernel = FALSE;
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self->residue_length = 0;
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self->next_ts = GST_CLOCK_TIME_NONE;
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self->next_off = GST_BUFFER_OFFSET_NONE;
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gst_pad_set_query_function (GST_BASE_TRANSFORM (self)->srcpad,
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audio_wsinclimit_query);
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gst_pad_set_query_type_function (GST_BASE_TRANSFORM (self)->srcpad,
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audio_wsinclimit_query_type);
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}
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#define DEFINE_PROCESS_FUNC(width,ctype) \
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static void \
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process_##width (GstAudioWSincLimit * self, g##ctype * src, g##ctype * dst, guint input_samples) \
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{ \
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gint kernel_length = self->kernel_length; \
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gint i, j, k, l; \
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gint channels = GST_AUDIO_FILTER (self)->format.channels; \
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gint res_start; \
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\
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/* convolution */ \
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for (i = 0; i < input_samples; i++) { \
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dst[i] = 0.0; \
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k = i % channels; \
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l = i / channels; \
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for (j = 0; j < kernel_length; j++) \
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if (l < j) \
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dst[i] += \
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self->residue[(kernel_length + l - j) * channels + \
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k] * self->kernel[j]; \
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else \
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dst[i] += src[(l - j) * channels + k] * self->kernel[j]; \
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} \
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\
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/* copy the tail of the current input buffer to the residue, while \
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* keeping parts of the residue if the input buffer is smaller than \
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* the kernel length */ \
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if (input_samples < kernel_length * channels) \
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res_start = kernel_length * channels - input_samples; \
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else \
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res_start = 0; \
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\
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for (i = 0; i < res_start; i++) \
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self->residue[i] = self->residue[i + input_samples]; \
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for (i = res_start; i < kernel_length * channels; i++) \
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self->residue[i] = src[input_samples - kernel_length * channels + i]; \
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\
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self->residue_length += kernel_length * channels - res_start; \
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if (self->residue_length > kernel_length * channels) \
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self->residue_length = kernel_length * channels; \
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}
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DEFINE_PROCESS_FUNC (32, float);
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DEFINE_PROCESS_FUNC (64, double);
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#undef DEFINE_PROCESS_FUNC
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static void
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audio_wsinclimit_build_kernel (GstAudioWSincLimit * self)
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{
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gint i = 0;
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gdouble sum = 0.0;
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gint len = 0;
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gdouble w;
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len = self->kernel_length;
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if (GST_AUDIO_FILTER (self)->format.rate == 0) {
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GST_DEBUG ("rate not set yet");
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return;
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}
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if (GST_AUDIO_FILTER (self)->format.channels == 0) {
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GST_DEBUG ("channels not set yet");
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return;
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}
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/* Clamp cutoff frequency between 0 and the nyquist frequency */
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self->cutoff =
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CLAMP (self->cutoff, 0.0, GST_AUDIO_FILTER (self)->format.rate / 2);
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GST_DEBUG ("audio_wsinclimit_: initializing filter kernel of length %d "
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"with cutoff %.2lf Hz "
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"for mode %s",
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len, self->cutoff,
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(self->mode == MODE_LOW_PASS) ? "low-pass" : "high-pass");
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/* fill the kernel */
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w = 2 * M_PI * (self->cutoff / GST_AUDIO_FILTER (self)->format.rate);
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if (self->kernel)
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g_free (self->kernel);
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self->kernel = g_new (gdouble, len);
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for (i = 0; i < len; ++i) {
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if (i == len / 2)
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self->kernel[i] = w;
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else
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self->kernel[i] = sin (w * (i - len / 2)) / (i - len / 2);
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/* windowing */
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if (self->window == WINDOW_HAMMING)
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self->kernel[i] *= (0.54 - 0.46 * cos (2 * M_PI * i / len));
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else
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self->kernel[i] *=
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(0.42 - 0.5 * cos (2 * M_PI * i / len) +
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0.08 * cos (4 * M_PI * i / len));
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}
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/* normalize for unity gain at DC */
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for (i = 0; i < len; ++i)
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sum += self->kernel[i];
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for (i = 0; i < len; ++i)
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self->kernel[i] /= sum;
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/* convert to highpass if specified */
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if (self->mode == MODE_HIGH_PASS) {
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for (i = 0; i < len; ++i)
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self->kernel[i] = -self->kernel[i];
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self->kernel[len / 2] += 1.0;
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}
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/* set up the residue memory space */
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if (!self->residue) {
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self->residue =
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g_new0 (gdouble, len * GST_AUDIO_FILTER (self)->format.channels);
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self->residue_length = 0;
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}
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self->have_kernel = TRUE;
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}
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static void
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audio_wsinclimit_push_residue (GstAudioWSincLimit * self)
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{
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GstBuffer *outbuf;
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GstFlowReturn res;
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gint rate = GST_AUDIO_FILTER (self)->format.rate;
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gint channels = GST_AUDIO_FILTER (self)->format.channels;
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gint outsize, outsamples;
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gint diffsize, diffsamples;
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guint8 *in, *out;
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/* Calculate the number of samples and their memory size that
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* should be pushed from the residue */
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outsamples = MIN (self->latency, self->residue_length / channels);
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outsize = outsamples * channels * (GST_AUDIO_FILTER (self)->format.width / 8);
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if (outsize == 0)
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return;
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/* Process the difference between latency and residue_length samples
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* to start at the actual data instead of starting at the zeros before
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* when we only got one buffer smaller than latency */
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diffsamples = self->latency - self->residue_length / channels;
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diffsize =
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diffsamples * channels * (GST_AUDIO_FILTER (self)->format.width / 8);
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if (diffsize > 0) {
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in = g_new0 (guint8, diffsize);
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out = g_new0 (guint8, diffsize);
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self->process (self, in, out, diffsamples * channels);
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g_free (in);
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g_free (out);
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}
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res = gst_pad_alloc_buffer (GST_BASE_TRANSFORM (self)->srcpad,
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GST_BUFFER_OFFSET_NONE, outsize,
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GST_PAD_CAPS (GST_BASE_TRANSFORM (self)->srcpad), &outbuf);
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if (G_UNLIKELY (res != GST_FLOW_OK)) {
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GST_WARNING_OBJECT (self, "failed allocating buffer of %d bytes", outsize);
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return;
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}
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/* Convolve the residue with zeros to get the actual remaining data */
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in = g_new0 (guint8, outsize);
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self->process (self, in, GST_BUFFER_DATA (outbuf), outsamples * channels);
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g_free (in);
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/* Set timestamp, offset, etc from the values we
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* saved when processing the regular buffers */
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if (GST_CLOCK_TIME_IS_VALID (self->next_ts))
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GST_BUFFER_TIMESTAMP (outbuf) = self->next_ts;
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else
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GST_BUFFER_TIMESTAMP (outbuf) = 0;
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GST_BUFFER_DURATION (outbuf) =
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gst_util_uint64_scale (outsamples, GST_SECOND, rate);
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self->next_ts += gst_util_uint64_scale (outsamples, GST_SECOND, rate);
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if (self->next_off != GST_BUFFER_OFFSET_NONE) {
|
|
GST_BUFFER_OFFSET (outbuf) = self->next_off;
|
|
GST_BUFFER_OFFSET_END (outbuf) = self->next_off + outsamples;
|
|
}
|
|
|
|
GST_DEBUG_OBJECT (self, "Pushing residue buffer of size %d with timestamp: %"
|
|
GST_TIME_FORMAT ", duration: %" GST_TIME_FORMAT ", offset: %lld,"
|
|
" offset_end: %lld, nsamples: %d", GST_BUFFER_SIZE (outbuf),
|
|
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf)),
|
|
GST_TIME_ARGS (GST_BUFFER_DURATION (outbuf)), GST_BUFFER_OFFSET (outbuf),
|
|
GST_BUFFER_OFFSET_END (outbuf), outsamples);
|
|
|
|
res = gst_pad_push (GST_BASE_TRANSFORM (self)->srcpad, outbuf);
|
|
|
|
if (G_UNLIKELY (res != GST_FLOW_OK)) {
|
|
GST_WARNING_OBJECT (self, "failed to push residue");
|
|
}
|
|
|
|
}
|
|
|
|
/* GstAudioFilter vmethod implementations */
|
|
|
|
/* get notified of caps and plug in the correct process function */
|
|
static gboolean
|
|
audio_wsinclimit_setup (GstAudioFilter * base, GstRingBufferSpec * format)
|
|
{
|
|
GstAudioWSincLimit *self = GST_AUDIO_WSINC_LIMIT (base);
|
|
|
|
gboolean ret = TRUE;
|
|
|
|
if (format->width == 32)
|
|
self->process = (GstAudioWSincLimitProcessFunc) process_32;
|
|
else if (format->width == 64)
|
|
self->process = (GstAudioWSincLimitProcessFunc) process_64;
|
|
else
|
|
ret = FALSE;
|
|
|
|
self->have_kernel = FALSE;
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
/* GstBaseTransform vmethod implementations */
|
|
|
|
static GstFlowReturn
|
|
audio_wsinclimit_transform (GstBaseTransform * base, GstBuffer * inbuf,
|
|
GstBuffer * outbuf)
|
|
{
|
|
GstAudioWSincLimit *self = GST_AUDIO_WSINC_LIMIT (base);
|
|
GstClockTime timestamp;
|
|
gint channels = GST_AUDIO_FILTER (self)->format.channels;
|
|
gint rate = GST_AUDIO_FILTER (self)->format.rate;
|
|
gint input_samples =
|
|
GST_BUFFER_SIZE (outbuf) / (GST_AUDIO_FILTER (self)->format.width / 8);
|
|
gint output_samples = input_samples;
|
|
gint diff;
|
|
|
|
/* FIXME: subdivide GST_BUFFER_SIZE into small chunks for smooth fades */
|
|
timestamp = GST_BUFFER_TIMESTAMP (outbuf);
|
|
if (GST_CLOCK_TIME_IS_VALID (timestamp))
|
|
gst_object_sync_values (G_OBJECT (self), timestamp);
|
|
|
|
if (!self->have_kernel)
|
|
audio_wsinclimit_build_kernel (self);
|
|
|
|
/* Reset the residue if already existing on discont buffers */
|
|
if (GST_BUFFER_IS_DISCONT (inbuf)) {
|
|
if (channels && self->residue)
|
|
memset (self->residue, 0, channels *
|
|
self->kernel_length * sizeof (gdouble));
|
|
self->residue_length = 0;
|
|
self->next_ts = GST_CLOCK_TIME_NONE;
|
|
self->next_off = GST_BUFFER_OFFSET_NONE;
|
|
}
|
|
|
|
/* Calculate the number of samples we can push out now without outputting
|
|
* kernel_length/2 zeros in the beginning */
|
|
diff = (self->kernel_length / 2) * channels - self->residue_length;
|
|
if (diff > 0)
|
|
output_samples -= diff;
|
|
|
|
self->process (self, GST_BUFFER_DATA (inbuf), GST_BUFFER_DATA (outbuf),
|
|
input_samples);
|
|
|
|
if (output_samples <= 0) {
|
|
/* Drop buffer and save original timestamp/offset for later use */
|
|
if (!GST_CLOCK_TIME_IS_VALID (self->next_ts)
|
|
&& GST_BUFFER_TIMESTAMP_IS_VALID (outbuf))
|
|
self->next_ts = GST_BUFFER_TIMESTAMP (outbuf);
|
|
if (self->next_off == GST_BUFFER_OFFSET_NONE
|
|
&& GST_BUFFER_OFFSET_IS_VALID (outbuf))
|
|
self->next_off = GST_BUFFER_OFFSET (outbuf);
|
|
return GST_BASE_TRANSFORM_FLOW_DROPPED;
|
|
} else if (output_samples < input_samples) {
|
|
/* First (probably partial) buffer after starting from
|
|
* a clean residue. Use stored timestamp/offset here */
|
|
if (GST_CLOCK_TIME_IS_VALID (self->next_ts))
|
|
GST_BUFFER_TIMESTAMP (outbuf) = self->next_ts;
|
|
|
|
if (self->next_off != GST_BUFFER_OFFSET_NONE) {
|
|
GST_BUFFER_OFFSET (outbuf) = self->next_off;
|
|
if (GST_BUFFER_OFFSET_END_IS_VALID (outbuf))
|
|
GST_BUFFER_OFFSET_END (outbuf) =
|
|
self->next_off + output_samples / channels;
|
|
} else {
|
|
/* We dropped no buffer, offset is valid, offset_end must be adjusted by diff */
|
|
if (GST_BUFFER_OFFSET_END_IS_VALID (outbuf))
|
|
GST_BUFFER_OFFSET_END (outbuf) -= diff / channels;
|
|
}
|
|
|
|
if (GST_BUFFER_DURATION_IS_VALID (outbuf))
|
|
GST_BUFFER_DURATION (outbuf) -=
|
|
gst_util_uint64_scale (diff, GST_SECOND, channels * rate);
|
|
|
|
GST_BUFFER_DATA (outbuf) +=
|
|
diff * (GST_AUDIO_FILTER (self)->format.width / 8);
|
|
GST_BUFFER_SIZE (outbuf) -=
|
|
diff * (GST_AUDIO_FILTER (self)->format.width / 8);
|
|
} else {
|
|
GstClockTime ts_latency =
|
|
gst_util_uint64_scale (self->latency, GST_SECOND, rate);
|
|
|
|
/* Normal buffer, adjust timestamp/offset/etc by latency */
|
|
if (GST_BUFFER_TIMESTAMP (outbuf) < ts_latency) {
|
|
GST_WARNING_OBJECT (self, "GST_BUFFER_TIMESTAMP (outbuf) < latency");
|
|
GST_BUFFER_TIMESTAMP (outbuf) = 0;
|
|
} else {
|
|
GST_BUFFER_TIMESTAMP (outbuf) -= ts_latency;
|
|
}
|
|
|
|
if (GST_BUFFER_OFFSET_IS_VALID (outbuf)) {
|
|
if (GST_BUFFER_OFFSET (outbuf) > self->latency) {
|
|
GST_BUFFER_OFFSET (outbuf) -= self->latency;
|
|
} else {
|
|
GST_WARNING_OBJECT (self, "GST_BUFFER_OFFSET (outbuf) < latency");
|
|
GST_BUFFER_OFFSET (outbuf) = 0;
|
|
}
|
|
}
|
|
|
|
if (GST_BUFFER_OFFSET_END_IS_VALID (outbuf)) {
|
|
if (GST_BUFFER_OFFSET_END (outbuf) > self->latency) {
|
|
GST_BUFFER_OFFSET_END (outbuf) -= self->latency;
|
|
} else {
|
|
GST_WARNING_OBJECT (self, "GST_BUFFER_OFFSET_END (outbuf) < latency");
|
|
GST_BUFFER_OFFSET_END (outbuf) = 0;
|
|
}
|
|
}
|
|
}
|
|
|
|
GST_DEBUG_OBJECT (self, "Pushing buffer of size %d with timestamp: %"
|
|
GST_TIME_FORMAT ", duration: %" GST_TIME_FORMAT ", offset: %lld,"
|
|
" offset_end: %lld, nsamples: %d", GST_BUFFER_SIZE (outbuf),
|
|
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf)),
|
|
GST_TIME_ARGS (GST_BUFFER_DURATION (outbuf)), GST_BUFFER_OFFSET (outbuf),
|
|
GST_BUFFER_OFFSET_END (outbuf), output_samples / channels);
|
|
|
|
self->next_ts = GST_BUFFER_TIMESTAMP (outbuf) + GST_BUFFER_DURATION (outbuf);
|
|
self->next_off = GST_BUFFER_OFFSET_END (outbuf);
|
|
|
|
return GST_FLOW_OK;
|
|
}
|
|
|
|
static gboolean
|
|
audio_wsinclimit_start (GstBaseTransform * base)
|
|
{
|
|
GstAudioWSincLimit *self = GST_AUDIO_WSINC_LIMIT (base);
|
|
gint channels = GST_AUDIO_FILTER (self)->format.channels;
|
|
|
|
/* Reset the residue if already existing */
|
|
if (channels && self->residue)
|
|
memset (self->residue, 0, channels *
|
|
self->kernel_length * sizeof (gdouble));
|
|
|
|
self->residue_length = 0;
|
|
self->next_ts = GST_CLOCK_TIME_NONE;
|
|
self->next_off = GST_BUFFER_OFFSET_NONE;
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
static gboolean
|
|
audio_wsinclimit_query (GstPad * pad, GstQuery * query)
|
|
{
|
|
GstAudioWSincLimit *self = GST_AUDIO_WSINC_LIMIT (gst_pad_get_parent (pad));
|
|
gboolean res = TRUE;
|
|
|
|
switch (GST_QUERY_TYPE (query)) {
|
|
case GST_QUERY_LATENCY:
|
|
{
|
|
GstClockTime min, max;
|
|
gboolean live;
|
|
guint64 latency;
|
|
GstPad *peer;
|
|
gint rate = GST_AUDIO_FILTER (self)->format.rate;
|
|
|
|
if ((peer = gst_pad_get_peer (GST_BASE_TRANSFORM (self)->sinkpad))) {
|
|
if ((res = gst_pad_query (peer, query))) {
|
|
gst_query_parse_latency (query, &live, &min, &max);
|
|
|
|
GST_DEBUG_OBJECT (self, "Peer latency: min %"
|
|
GST_TIME_FORMAT " max %" GST_TIME_FORMAT,
|
|
GST_TIME_ARGS (min), GST_TIME_ARGS (max));
|
|
|
|
/* add our own latency */
|
|
latency =
|
|
(rate != 0) ? gst_util_uint64_scale (self->latency, GST_SECOND,
|
|
rate) : 0;
|
|
|
|
GST_DEBUG_OBJECT (self, "Our latency: %"
|
|
GST_TIME_FORMAT, GST_TIME_ARGS (latency));
|
|
|
|
min += latency;
|
|
if (max != GST_CLOCK_TIME_NONE)
|
|
max += latency;
|
|
|
|
GST_DEBUG_OBJECT (self, "Calculated total latency : min %"
|
|
GST_TIME_FORMAT " max %" GST_TIME_FORMAT,
|
|
GST_TIME_ARGS (min), GST_TIME_ARGS (max));
|
|
|
|
gst_query_set_latency (query, live, min, max);
|
|
}
|
|
gst_object_unref (peer);
|
|
}
|
|
break;
|
|
}
|
|
default:
|
|
res = gst_pad_query_default (pad, query);
|
|
break;
|
|
}
|
|
gst_object_unref (self);
|
|
return res;
|
|
}
|
|
|
|
static const GstQueryType *
|
|
audio_wsinclimit_query_type (GstPad * pad)
|
|
{
|
|
static const GstQueryType types[] = {
|
|
GST_QUERY_LATENCY,
|
|
0
|
|
};
|
|
|
|
return types;
|
|
}
|
|
|
|
static gboolean
|
|
audio_wsinclimit_event (GstBaseTransform * base, GstEvent * event)
|
|
{
|
|
GstAudioWSincLimit *self = GST_AUDIO_WSINC_LIMIT (base);
|
|
|
|
switch (GST_EVENT_TYPE (event)) {
|
|
case GST_EVENT_EOS:
|
|
audio_wsinclimit_push_residue (self);
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
|
|
return GST_BASE_TRANSFORM_CLASS (parent_class)->event (base, event);
|
|
}
|
|
|
|
static void
|
|
audio_wsinclimit_set_property (GObject * object, guint prop_id,
|
|
const GValue * value, GParamSpec * pspec)
|
|
{
|
|
GstAudioWSincLimit *self = GST_AUDIO_WSINC_LIMIT (object);
|
|
|
|
g_return_if_fail (GST_IS_AUDIO_WSINC_LIMIT (self));
|
|
|
|
switch (prop_id) {
|
|
case PROP_LENGTH:{
|
|
gint val;
|
|
|
|
GST_BASE_TRANSFORM_LOCK (self);
|
|
val = g_value_get_int (value);
|
|
if (val % 2 == 0)
|
|
val++;
|
|
|
|
if (val != self->kernel_length) {
|
|
if (self->residue) {
|
|
audio_wsinclimit_push_residue (self);
|
|
g_free (self->residue);
|
|
self->residue = NULL;
|
|
}
|
|
self->kernel_length = val;
|
|
self->latency = val / 2;
|
|
audio_wsinclimit_build_kernel (self);
|
|
gst_element_post_message (GST_ELEMENT (self),
|
|
gst_message_new_latency (GST_OBJECT (self)));
|
|
}
|
|
GST_BASE_TRANSFORM_UNLOCK (self);
|
|
break;
|
|
}
|
|
case PROP_FREQUENCY:
|
|
GST_BASE_TRANSFORM_LOCK (self);
|
|
self->cutoff = g_value_get_float (value);
|
|
audio_wsinclimit_build_kernel (self);
|
|
GST_BASE_TRANSFORM_UNLOCK (self);
|
|
break;
|
|
case PROP_MODE:
|
|
GST_BASE_TRANSFORM_LOCK (self);
|
|
self->mode = g_value_get_enum (value);
|
|
audio_wsinclimit_build_kernel (self);
|
|
GST_BASE_TRANSFORM_UNLOCK (self);
|
|
break;
|
|
case PROP_WINDOW:
|
|
GST_BASE_TRANSFORM_LOCK (self);
|
|
self->window = g_value_get_enum (value);
|
|
audio_wsinclimit_build_kernel (self);
|
|
GST_BASE_TRANSFORM_UNLOCK (self);
|
|
break;
|
|
default:
|
|
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
|
|
break;
|
|
}
|
|
}
|
|
|
|
static void
|
|
audio_wsinclimit_get_property (GObject * object, guint prop_id, GValue * value,
|
|
GParamSpec * pspec)
|
|
{
|
|
GstAudioWSincLimit *self = GST_AUDIO_WSINC_LIMIT (object);
|
|
|
|
switch (prop_id) {
|
|
case PROP_LENGTH:
|
|
g_value_set_int (value, self->kernel_length);
|
|
break;
|
|
case PROP_FREQUENCY:
|
|
g_value_set_float (value, self->cutoff);
|
|
break;
|
|
case PROP_MODE:
|
|
g_value_set_enum (value, self->mode);
|
|
break;
|
|
case PROP_WINDOW:
|
|
g_value_set_enum (value, self->window);
|
|
break;
|
|
default:
|
|
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
|
|
break;
|
|
}
|
|
}
|