mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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43576fb0cf
Remove some redundant calculations, move comparisions out of inner loops, etc. This makes the convolution about 3 (!) times faster but processing time is of course still proportional to the filter size.
564 lines
18 KiB
C
564 lines
18 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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* 2007-2009 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., 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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* 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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#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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/* FIXME: Remove this once we depend on gst-plugins-base 0.10.26 */
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#ifndef GST_AUDIO_FILTER_CAST
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#define GST_AUDIO_FILTER_CAST(obj) ((GstAudioFilter *) (obj))
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#endif
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#include "audiofxbasefirfilter.h"
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#define GST_CAT_DEFAULT gst_audio_fx_base_fir_filter_debug
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GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
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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 (gst_audio_fx_base_fir_filter_debug, "audiofxbasefirfilter", 0, \
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"FIR filter base class");
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GST_BOILERPLATE_FULL (GstAudioFXBaseFIRFilter, gst_audio_fx_base_fir_filter,
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GstAudioFilter, GST_TYPE_AUDIO_FILTER, DEBUG_INIT);
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static GstFlowReturn gst_audio_fx_base_fir_filter_transform (GstBaseTransform *
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base, GstBuffer * inbuf, GstBuffer * outbuf);
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static gboolean gst_audio_fx_base_fir_filter_start (GstBaseTransform * base);
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static gboolean gst_audio_fx_base_fir_filter_stop (GstBaseTransform * base);
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static gboolean gst_audio_fx_base_fir_filter_event (GstBaseTransform * base,
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GstEvent * event);
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static gboolean gst_audio_fx_base_fir_filter_setup (GstAudioFilter * base,
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GstRingBufferSpec * format);
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static gboolean gst_audio_fx_base_fir_filter_query (GstPad * pad,
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GstQuery * query);
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static const GstQueryType *gst_audio_fx_base_fir_filter_query_type (GstPad *
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pad);
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/* Element class */
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static void
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gst_audio_fx_base_fir_filter_dispose (GObject * object)
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{
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GstAudioFXBaseFIRFilter *self = GST_AUDIO_FX_BASE_FIR_FILTER (object);
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if (self->buffer) {
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g_free (self->buffer);
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self->buffer = 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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gst_audio_fx_base_fir_filter_base_init (gpointer g_class)
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{
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GstCaps *caps;
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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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gst_audio_fx_base_fir_filter_class_init (GstAudioFXBaseFIRFilterClass * klass)
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{
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GObjectClass *gobject_class = (GObjectClass *) klass;
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GstBaseTransformClass *trans_class = (GstBaseTransformClass *) klass;
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GstAudioFilterClass *filter_class = (GstAudioFilterClass *) klass;
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gobject_class->dispose = gst_audio_fx_base_fir_filter_dispose;
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trans_class->transform =
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GST_DEBUG_FUNCPTR (gst_audio_fx_base_fir_filter_transform);
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trans_class->start = GST_DEBUG_FUNCPTR (gst_audio_fx_base_fir_filter_start);
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trans_class->stop = GST_DEBUG_FUNCPTR (gst_audio_fx_base_fir_filter_stop);
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trans_class->event = GST_DEBUG_FUNCPTR (gst_audio_fx_base_fir_filter_event);
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filter_class->setup = GST_DEBUG_FUNCPTR (gst_audio_fx_base_fir_filter_setup);
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}
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static void
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gst_audio_fx_base_fir_filter_init (GstAudioFXBaseFIRFilter * self,
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GstAudioFXBaseFIRFilterClass * g_class)
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{
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self->kernel = NULL;
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self->buffer = NULL;
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self->start_ts = GST_CLOCK_TIME_NONE;
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self->start_off = GST_BUFFER_OFFSET_NONE;
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self->nsamples = 0;
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gst_pad_set_query_function (GST_BASE_TRANSFORM (self)->srcpad,
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gst_audio_fx_base_fir_filter_query);
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gst_pad_set_query_type_function (GST_BASE_TRANSFORM (self)->srcpad,
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gst_audio_fx_base_fir_filter_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 (GstAudioFXBaseFIRFilter * self, const 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_CAST (self)->format.channels; \
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gint res_start; \
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gint from_input; \
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gint off; \
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gdouble *buffer = self->buffer; \
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gdouble *kernel = self->kernel; \
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guint buffer_length = self->buffer_length; \
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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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from_input = MIN (l, kernel_length-1); \
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off = l * channels + k; \
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for (j = 0; j <= from_input; j++) { \
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dst[i] += src[off] * kernel[j]; \
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off -= channels; \
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} \
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/* j == from_input && off == (l - j) * channels + k */ \
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off += kernel_length * channels; \
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for (; j < kernel_length; j++) { \
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dst[i] += buffer[off] * kernel[j]; \
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off -= channels; \
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} \
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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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/* from now on take kernel length as length over all channels */ \
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kernel_length *= channels; \
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if (input_samples < kernel_length) \
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res_start = kernel_length - 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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buffer[i] = buffer[i + input_samples]; \
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/* i == res_start */ \
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for (; i < kernel_length; i++) \
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buffer[i] = src[input_samples - kernel_length + i]; \
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\
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self->buffer_fill += kernel_length - res_start; \
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if (self->buffer_fill > kernel_length) \
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self->buffer_fill = kernel_length; \
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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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void
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gst_audio_fx_base_fir_filter_push_residue (GstAudioFXBaseFIRFilter * 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_CAST (self)->format.rate;
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gint channels = GST_AUDIO_FILTER_CAST (self)->format.channels;
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gint outsize, outsamples;
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gint diffsize, diffsamples;
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gint width = GST_AUDIO_FILTER_CAST (self)->format.width / 8;
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guint8 *in, *out;
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if (channels == 0 || rate == 0) {
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self->buffer_fill = 0;
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return;
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}
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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->buffer_fill / channels);
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outsize = outsamples * channels * width;
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if (outsize == 0) {
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self->buffer_fill = 0;
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return;
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}
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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->buffer_fill / channels;
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if (diffsamples > 0) {
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diffsize = diffsamples * channels * width;
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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_CAST (self)->srcpad,
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GST_BUFFER_OFFSET_NONE, outsize,
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GST_PAD_CAPS (GST_BASE_TRANSFORM_CAST (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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self->buffer_fill = 0;
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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->start_ts))
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GST_BUFFER_TIMESTAMP (outbuf) = self->start_ts;
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else
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GST_BUFFER_TIMESTAMP (outbuf) = 0;
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GST_BUFFER_TIMESTAMP (outbuf) +=
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gst_util_uint64_scale_round (self->nsamples, GST_SECOND, rate);
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GST_BUFFER_DURATION (outbuf) =
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gst_util_uint64_scale_round (outsamples, GST_SECOND, rate);
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if (self->start_off != GST_BUFFER_OFFSET_NONE) {
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GST_BUFFER_OFFSET (outbuf) = self->start_off + self->nsamples;
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GST_BUFFER_OFFSET_END (outbuf) = GST_BUFFER_OFFSET (outbuf) + outsamples;
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}
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self->nsamples += outsamples;
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GST_DEBUG_OBJECT (self, "Pushing residue buffer of size %d with timestamp: %"
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GST_TIME_FORMAT ", duration: %" GST_TIME_FORMAT ", offset: %"
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G_GUINT64_FORMAT ", offset_end: %" G_GUINT64_FORMAT ", nsamples: %d",
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GST_BUFFER_SIZE (outbuf), GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf)),
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GST_TIME_ARGS (GST_BUFFER_DURATION (outbuf)), GST_BUFFER_OFFSET (outbuf),
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GST_BUFFER_OFFSET_END (outbuf), outsamples);
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res = gst_pad_push (GST_BASE_TRANSFORM_CAST (self)->srcpad, outbuf);
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if (G_UNLIKELY (res != GST_FLOW_OK)) {
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GST_WARNING_OBJECT (self, "failed to push residue");
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}
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self->buffer_fill = 0;
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}
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/* GstAudioFilter vmethod implementations */
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/* get notified of caps and plug in the correct process function */
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static gboolean
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gst_audio_fx_base_fir_filter_setup (GstAudioFilter * base,
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GstRingBufferSpec * format)
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{
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GstAudioFXBaseFIRFilter *self = GST_AUDIO_FX_BASE_FIR_FILTER (base);
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gboolean ret = TRUE;
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if (self->buffer) {
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gst_audio_fx_base_fir_filter_push_residue (self);
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g_free (self->buffer);
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self->buffer = NULL;
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self->buffer_fill = 0;
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self->start_ts = GST_CLOCK_TIME_NONE;
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self->start_off = GST_BUFFER_OFFSET_NONE;
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self->nsamples = 0;
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}
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if (format->width == 32)
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self->process = (GstAudioFXBaseFIRFilterProcessFunc) process_32;
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else if (format->width == 64)
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self->process = (GstAudioFXBaseFIRFilterProcessFunc) process_64;
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else
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ret = FALSE;
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return TRUE;
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}
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/* GstBaseTransform vmethod implementations */
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static GstFlowReturn
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gst_audio_fx_base_fir_filter_transform (GstBaseTransform * base,
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GstBuffer * inbuf, GstBuffer * outbuf)
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{
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GstAudioFXBaseFIRFilter *self = GST_AUDIO_FX_BASE_FIR_FILTER (base);
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GstClockTime timestamp, expected_timestamp;
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gint channels = GST_AUDIO_FILTER_CAST (self)->format.channels;
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gint rate = GST_AUDIO_FILTER_CAST (self)->format.rate;
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gint width = GST_AUDIO_FILTER_CAST (self)->format.width / 8;
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gint input_samples = (GST_BUFFER_SIZE (outbuf) / width) / channels;
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gint output_samples = input_samples;
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gint diff = 0;
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timestamp = GST_BUFFER_TIMESTAMP (outbuf);
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if (!GST_CLOCK_TIME_IS_VALID (timestamp)
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&& !GST_CLOCK_TIME_IS_VALID (self->start_ts)) {
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GST_ERROR_OBJECT (self, "Invalid timestamp");
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return GST_FLOW_ERROR;
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}
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gst_object_sync_values (G_OBJECT (self), timestamp);
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g_return_val_if_fail (self->kernel != NULL, GST_FLOW_ERROR);
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g_return_val_if_fail (channels != 0, GST_FLOW_ERROR);
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if (!self->buffer)
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self->buffer = g_new0 (gdouble, self->kernel_length * channels);
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if (GST_CLOCK_TIME_IS_VALID (self->start_ts))
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expected_timestamp =
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self->start_ts + gst_util_uint64_scale_round (self->nsamples,
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GST_SECOND, rate);
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else
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expected_timestamp = GST_CLOCK_TIME_NONE;
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/* Reset the residue if already existing on discont buffers */
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if (GST_BUFFER_IS_DISCONT (inbuf)
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|| (GST_CLOCK_TIME_IS_VALID (expected_timestamp)
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&& timestamp - gst_util_uint64_scale_round (MIN (self->latency,
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self->buffer_fill / channels), GST_SECOND,
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rate) - expected_timestamp > 5 * GST_MSECOND)) {
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GST_DEBUG_OBJECT (self, "Discontinuity detected - flushing");
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if (GST_CLOCK_TIME_IS_VALID (expected_timestamp))
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gst_audio_fx_base_fir_filter_push_residue (self);
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self->buffer_fill = 0;
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expected_timestamp = self->start_ts = timestamp;
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self->start_off = GST_BUFFER_OFFSET (inbuf);
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self->nsamples = 0;
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} else if (!GST_CLOCK_TIME_IS_VALID (self->start_ts)) {
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expected_timestamp = self->start_ts = timestamp;
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self->start_off = GST_BUFFER_OFFSET (inbuf);
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}
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/* Calculate the number of samples we can push out now without outputting
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* latency zeros in the beginning */
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diff = self->latency - self->buffer_fill / channels;
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if (diff > 0)
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output_samples -= diff;
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self->process (self, GST_BUFFER_DATA (inbuf), GST_BUFFER_DATA (outbuf),
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input_samples * channels);
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if (output_samples <= 0) {
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return GST_BASE_TRANSFORM_FLOW_DROPPED;
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}
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GST_BUFFER_TIMESTAMP (outbuf) = expected_timestamp;
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GST_BUFFER_DURATION (outbuf) =
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gst_util_uint64_scale_round (output_samples, GST_SECOND, rate);
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if (self->start_off != GST_BUFFER_OFFSET_NONE) {
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GST_BUFFER_OFFSET (outbuf) = self->start_off + self->nsamples;
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GST_BUFFER_OFFSET_END (outbuf) = self->start_off + output_samples;
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} else {
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GST_BUFFER_OFFSET (outbuf) = GST_BUFFER_OFFSET_NONE;
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GST_BUFFER_OFFSET_END (outbuf) = GST_BUFFER_OFFSET_NONE;
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}
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if (output_samples < input_samples) {
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GST_BUFFER_DATA (outbuf) += diff * width;
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GST_BUFFER_SIZE (outbuf) -= diff * width;
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}
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self->nsamples += output_samples;
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GST_DEBUG_OBJECT (self, "Pushing buffer of size %d with timestamp: %"
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GST_TIME_FORMAT ", duration: %" GST_TIME_FORMAT ", offset: %"
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G_GUINT64_FORMAT ", offset_end: %" G_GUINT64_FORMAT ", nsamples: %d",
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GST_BUFFER_SIZE (outbuf), GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf)),
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GST_TIME_ARGS (GST_BUFFER_DURATION (outbuf)), GST_BUFFER_OFFSET (outbuf),
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GST_BUFFER_OFFSET_END (outbuf), output_samples);
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return GST_FLOW_OK;
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}
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static gboolean
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gst_audio_fx_base_fir_filter_start (GstBaseTransform * base)
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{
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GstAudioFXBaseFIRFilter *self = GST_AUDIO_FX_BASE_FIR_FILTER (base);
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self->buffer_fill = 0;
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self->start_ts = GST_CLOCK_TIME_NONE;
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self->start_off = GST_BUFFER_OFFSET_NONE;
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self->nsamples = 0;
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return TRUE;
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}
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static gboolean
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gst_audio_fx_base_fir_filter_stop (GstBaseTransform * base)
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{
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GstAudioFXBaseFIRFilter *self = GST_AUDIO_FX_BASE_FIR_FILTER (base);
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g_free (self->buffer);
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self->buffer = NULL;
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return TRUE;
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}
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static gboolean
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gst_audio_fx_base_fir_filter_query (GstPad * pad, GstQuery * query)
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{
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GstAudioFXBaseFIRFilter *self =
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GST_AUDIO_FX_BASE_FIR_FILTER (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 (rate == 0) {
|
|
res = FALSE;
|
|
} else 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 =
|
|
gst_util_uint64_scale_round (self->latency, GST_SECOND, rate);
|
|
|
|
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 *
|
|
gst_audio_fx_base_fir_filter_query_type (GstPad * pad)
|
|
{
|
|
static const GstQueryType types[] = {
|
|
GST_QUERY_LATENCY,
|
|
0
|
|
};
|
|
|
|
return types;
|
|
}
|
|
|
|
static gboolean
|
|
gst_audio_fx_base_fir_filter_event (GstBaseTransform * base, GstEvent * event)
|
|
{
|
|
GstAudioFXBaseFIRFilter *self = GST_AUDIO_FX_BASE_FIR_FILTER (base);
|
|
|
|
switch (GST_EVENT_TYPE (event)) {
|
|
case GST_EVENT_EOS:
|
|
gst_audio_fx_base_fir_filter_push_residue (self);
|
|
self->start_ts = GST_CLOCK_TIME_NONE;
|
|
self->start_off = GST_BUFFER_OFFSET_NONE;
|
|
self->nsamples = 0;
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
|
|
return GST_BASE_TRANSFORM_CLASS (parent_class)->event (base, event);
|
|
}
|
|
|
|
void
|
|
gst_audio_fx_base_fir_filter_set_kernel (GstAudioFXBaseFIRFilter * self,
|
|
gdouble * kernel, guint kernel_length, guint64 latency)
|
|
{
|
|
g_return_if_fail (kernel != NULL);
|
|
g_return_if_fail (self != NULL);
|
|
|
|
GST_BASE_TRANSFORM_LOCK (self);
|
|
if (self->buffer) {
|
|
gst_audio_fx_base_fir_filter_push_residue (self);
|
|
self->start_ts = GST_CLOCK_TIME_NONE;
|
|
self->start_off = GST_BUFFER_OFFSET_NONE;
|
|
self->nsamples = 0;
|
|
self->buffer_fill = 0;
|
|
}
|
|
|
|
g_free (self->kernel);
|
|
g_free (self->buffer);
|
|
|
|
self->kernel = kernel;
|
|
self->kernel_length = kernel_length;
|
|
|
|
if (GST_AUDIO_FILTER (self)->format.channels) {
|
|
self->buffer =
|
|
g_new0 (gdouble,
|
|
kernel_length * GST_AUDIO_FILTER (self)->format.channels);
|
|
self->buffer_fill = 0;
|
|
}
|
|
|
|
if (self->latency != latency) {
|
|
self->latency = latency;
|
|
gst_element_post_message (GST_ELEMENT (self),
|
|
gst_message_new_latency (GST_OBJECT (self)));
|
|
}
|
|
|
|
GST_BASE_TRANSFORM_UNLOCK (self);
|
|
}
|