+2009-01-11 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/audiofx/Makefile.am:
+ * gst/audiofx/audiofxbasefirfilter.c:
+ (gst_audio_fx_base_fir_filter_dispose),
+ (gst_audio_fx_base_fir_filter_base_init),
+ (gst_audio_fx_base_fir_filter_class_init),
+ (gst_audio_fx_base_fir_filter_init),
+ (gst_audio_fx_base_fir_filter_push_residue),
+ (gst_audio_fx_base_fir_filter_setup),
+ (gst_audio_fx_base_fir_filter_transform),
+ (gst_audio_fx_base_fir_filter_start),
+ (gst_audio_fx_base_fir_filter_stop),
+ (gst_audio_fx_base_fir_filter_query),
+ (gst_audio_fx_base_fir_filter_query_type),
+ (gst_audio_fx_base_fir_filter_event),
+ (gst_audio_fx_base_fir_filter_set_kernel):
+ * gst/audiofx/audiofxbasefirfilter.h:
+ * gst/audiofx/audiofxbaseiirfilter.c:
+ Implement a base class for generic audio FIR filters.
+
+ * gst/audiofx/audiowsincband.c:
+ (gst_gst_audio_wsincband_mode_get_type),
+ (gst_gst_audio_wsincband_window_get_type),
+ (gst_audio_wsincband_base_init), (gst_audio_wsincband_class_init),
+ (gst_audio_wsincband_init), (gst_audio_wsincband_build_kernel),
+ (gst_audio_wsincband_setup), (gst_audio_wsincband_set_property),
+ (gst_audio_wsincband_get_property):
+ * gst/audiofx/audiowsincband.h:
+ * gst/audiofx/audiowsinclimit.c:
+ (gst_audio_wsinclimit_mode_get_type),
+ (gst_audio_wsinclimit_window_get_type),
+ (gst_audio_wsinclimit_base_init),
+ (gst_audio_wsinclimit_class_init), (gst_audio_wsinclimit_init),
+ (gst_audio_wsinclimit_build_kernel), (gst_audio_wsinclimit_setup),
+ (gst_audio_wsinclimit_set_property),
+ (gst_audio_wsinclimit_get_property):
+ * gst/audiofx/audiowsinclimit.h:
+ * tests/check/elements/audiowsincband.c: (GST_START_TEST):
+ * tests/check/elements/audiowsinclimit.c: (GST_START_TEST):
+ Use this new base class for audiowsincband and audiowsinclimit.
+ Also cleanup both elements.
+
2009-01-08 Michael Smith <msmith@songbirdnest.com>
* gst/qtdemux/qtdemux.c:
audiofxbaseiirfilter.c \
audiocheblimit.c \
audiochebband.c \
+ audiofxbasefirfilter.c \
audiowsincband.c \
audiowsinclimit.c
audiofxbaseiirfilter.h \
audiocheblimit.h \
audiochebband.h \
+ audiofxbasefirfilter.h \
audiowsincband.h \
audiowsinclimit.h \
math_compat.h
--- /dev/null
+/* -*- c-basic-offset: 2 -*-
+ *
+ * GStreamer
+ * Copyright (C) 1999-2001 Erik Walthinsen <omega@cse.ogi.edu>
+ * 2006 Dreamlab Technologies Ltd. <mathis.hofer@dreamlab.net>
+ * 2007-2009 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
+ * Boston, MA 02111-1307, USA.
+ *
+ *
+ * TODO: - Implement the convolution in place, probably only makes sense
+ * when using FFT convolution as currently the convolution itself
+ * is probably the bottleneck
+ * - Maybe allow cascading the filter to get a better stopband attenuation.
+ * Can be done by convolving a filter kernel with itself
+ */
+
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+
+#include <string.h>
+#include <math.h>
+#include <gst/gst.h>
+#include <gst/audio/gstaudiofilter.h>
+#include <gst/controller/gstcontroller.h>
+
+#include "audiofxbasefirfilter.h"
+
+#define GST_CAT_DEFAULT gst_audio_fx_base_fir_filter_debug
+GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
+
+#define ALLOWED_CAPS \
+ "audio/x-raw-float, " \
+ " width = (int) { 32, 64 }, " \
+ " endianness = (int) BYTE_ORDER, " \
+ " rate = (int) [ 1, MAX ], " \
+ " channels = (int) [ 1, MAX ]"
+
+#define DEBUG_INIT(bla) \
+ GST_DEBUG_CATEGORY_INIT (gst_audio_fx_base_fir_filter_debug, "audiofxbasefirfilter", 0, \
+ "FIR filter base class");
+
+GST_BOILERPLATE_FULL (GstAudioFXBaseFIRFilter, gst_audio_fx_base_fir_filter,
+ GstAudioFilter, GST_TYPE_AUDIO_FILTER, DEBUG_INIT);
+
+static GstFlowReturn gst_audio_fx_base_fir_filter_transform (GstBaseTransform *
+ base, GstBuffer * inbuf, GstBuffer * outbuf);
+static gboolean gst_audio_fx_base_fir_filter_start (GstBaseTransform * base);
+static gboolean gst_audio_fx_base_fir_filter_stop (GstBaseTransform * base);
+static gboolean gst_audio_fx_base_fir_filter_event (GstBaseTransform * base,
+ GstEvent * event);
+static gboolean gst_audio_fx_base_fir_filter_setup (GstAudioFilter * base,
+ GstRingBufferSpec * format);
+
+static gboolean gst_audio_fx_base_fir_filter_query (GstPad * pad,
+ GstQuery * query);
+static const GstQueryType *gst_audio_fx_base_fir_filter_query_type (GstPad *
+ pad);
+
+/* Element class */
+
+static void
+gst_audio_fx_base_fir_filter_dispose (GObject * object)
+{
+ GstAudioFXBaseFIRFilter *self = GST_AUDIO_FX_BASE_FIR_FILTER (object);
+
+ if (self->residue) {
+ g_free (self->residue);
+ self->residue = NULL;
+ }
+
+ if (self->kernel) {
+ g_free (self->kernel);
+ self->kernel = NULL;
+ }
+
+ G_OBJECT_CLASS (parent_class)->dispose (object);
+}
+
+static void
+gst_audio_fx_base_fir_filter_base_init (gpointer g_class)
+{
+ GstCaps *caps;
+
+ caps = gst_caps_from_string (ALLOWED_CAPS);
+ gst_audio_filter_class_add_pad_templates (GST_AUDIO_FILTER_CLASS (g_class),
+ caps);
+ gst_caps_unref (caps);
+}
+
+static void
+gst_audio_fx_base_fir_filter_class_init (GstAudioFXBaseFIRFilterClass * klass)
+{
+ GObjectClass *gobject_class = (GObjectClass *) klass;
+ GstBaseTransformClass *trans_class = (GstBaseTransformClass *) klass;
+ GstAudioFilterClass *filter_class = (GstAudioFilterClass *) klass;
+
+ gobject_class->dispose = gst_audio_fx_base_fir_filter_dispose;
+
+ trans_class->transform =
+ GST_DEBUG_FUNCPTR (gst_audio_fx_base_fir_filter_transform);
+ trans_class->start = GST_DEBUG_FUNCPTR (gst_audio_fx_base_fir_filter_start);
+ trans_class->stop = GST_DEBUG_FUNCPTR (gst_audio_fx_base_fir_filter_stop);
+ trans_class->event = GST_DEBUG_FUNCPTR (gst_audio_fx_base_fir_filter_event);
+ filter_class->setup = GST_DEBUG_FUNCPTR (gst_audio_fx_base_fir_filter_setup);
+}
+
+static void
+gst_audio_fx_base_fir_filter_init (GstAudioFXBaseFIRFilter * self,
+ GstAudioFXBaseFIRFilterClass * g_class)
+{
+ self->kernel = NULL;
+ self->residue = NULL;
+
+ self->next_ts = GST_CLOCK_TIME_NONE;
+ self->next_off = GST_BUFFER_OFFSET_NONE;
+
+ gst_pad_set_query_function (GST_BASE_TRANSFORM (self)->srcpad,
+ gst_audio_fx_base_fir_filter_query);
+ gst_pad_set_query_type_function (GST_BASE_TRANSFORM (self)->srcpad,
+ gst_audio_fx_base_fir_filter_query_type);
+}
+
+#define DEFINE_PROCESS_FUNC(width,ctype) \
+static void \
+process_##width (GstAudioFXBaseFIRFilter * self, g##ctype * src, g##ctype * dst, guint input_samples) \
+{ \
+ gint kernel_length = self->kernel_length; \
+ gint i, j, k, l; \
+ gint channels = GST_AUDIO_FILTER (self)->format.channels; \
+ gint res_start; \
+ \
+ /* convolution */ \
+ for (i = 0; i < input_samples; i++) { \
+ dst[i] = 0.0; \
+ k = i % channels; \
+ l = i / channels; \
+ for (j = 0; j < kernel_length; j++) \
+ if (l < j) \
+ dst[i] += \
+ self->residue[(kernel_length + l - j) * channels + \
+ k] * self->kernel[j]; \
+ else \
+ dst[i] += src[(l - j) * channels + k] * self->kernel[j]; \
+ } \
+ \
+ /* copy the tail of the current input buffer to the residue, while \
+ * keeping parts of the residue if the input buffer is smaller than \
+ * the kernel length */ \
+ if (input_samples < kernel_length * channels) \
+ res_start = kernel_length * channels - input_samples; \
+ else \
+ res_start = 0; \
+ \
+ for (i = 0; i < res_start; i++) \
+ self->residue[i] = self->residue[i + input_samples]; \
+ for (i = res_start; i < kernel_length * channels; i++) \
+ self->residue[i] = src[input_samples - kernel_length * channels + i]; \
+ \
+ self->residue_length += kernel_length * channels - res_start; \
+ if (self->residue_length > kernel_length * channels) \
+ self->residue_length = kernel_length * channels; \
+}
+
+DEFINE_PROCESS_FUNC (32, float);
+DEFINE_PROCESS_FUNC (64, double);
+
+#undef DEFINE_PROCESS_FUNC
+
+void
+gst_audio_fx_base_fir_filter_push_residue (GstAudioFXBaseFIRFilter * self)
+{
+ GstBuffer *outbuf;
+ GstFlowReturn res;
+ gint rate = GST_AUDIO_FILTER (self)->format.rate;
+ gint channels = GST_AUDIO_FILTER (self)->format.channels;
+ gint outsize, outsamples;
+ gint diffsize, diffsamples;
+ guint8 *in, *out;
+
+ if (channels == 0 || rate == 0) {
+ self->residue_length = 0;
+ return;
+ }
+
+ /* Calculate the number of samples and their memory size that
+ * should be pushed from the residue */
+ outsamples = MIN (self->latency, self->residue_length / channels);
+ outsize = outsamples * channels * (GST_AUDIO_FILTER (self)->format.width / 8);
+ if (outsize == 0) {
+ self->residue_length = 0;
+ return;
+ }
+
+ /* Process the difference between latency and residue_length samples
+ * to start at the actual data instead of starting at the zeros before
+ * when we only got one buffer smaller than latency */
+ diffsamples = self->latency - self->residue_length / channels;
+ diffsize =
+ diffsamples * channels * (GST_AUDIO_FILTER (self)->format.width / 8);
+ if (diffsize > 0) {
+ in = g_new0 (guint8, diffsize);
+ out = g_new0 (guint8, diffsize);
+ self->process (self, in, out, diffsamples * channels);
+ g_free (in);
+ g_free (out);
+ }
+
+ res = gst_pad_alloc_buffer (GST_BASE_TRANSFORM (self)->srcpad,
+ GST_BUFFER_OFFSET_NONE, outsize,
+ GST_PAD_CAPS (GST_BASE_TRANSFORM (self)->srcpad), &outbuf);
+
+ if (G_UNLIKELY (res != GST_FLOW_OK)) {
+ GST_WARNING_OBJECT (self, "failed allocating buffer of %d bytes", outsize);
+ self->residue_length = 0;
+ return;
+ }
+
+ /* Convolve the residue with zeros to get the actual remaining data */
+ in = g_new0 (guint8, outsize);
+ self->process (self, in, GST_BUFFER_DATA (outbuf), outsamples * channels);
+ g_free (in);
+
+ /* Set timestamp, offset, etc from the values we
+ * saved when processing the regular buffers */
+ if (GST_CLOCK_TIME_IS_VALID (self->next_ts))
+ GST_BUFFER_TIMESTAMP (outbuf) = self->next_ts;
+ else
+ GST_BUFFER_TIMESTAMP (outbuf) = 0;
+ GST_BUFFER_DURATION (outbuf) =
+ gst_util_uint64_scale (outsamples, GST_SECOND, rate);
+ self->next_ts += gst_util_uint64_scale (outsamples, GST_SECOND, rate);
+
+ if (self->next_off != GST_BUFFER_OFFSET_NONE) {
+ GST_BUFFER_OFFSET (outbuf) = self->next_off;
+ GST_BUFFER_OFFSET_END (outbuf) = self->next_off + outsamples;
+ self->next_off = GST_BUFFER_OFFSET_END (outbuf);
+ }
+
+ 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");
+ }
+
+ self->residue_length = 0;
+}
+
+/* GstAudioFilter vmethod implementations */
+
+/* get notified of caps and plug in the correct process function */
+static gboolean
+gst_audio_fx_base_fir_filter_setup (GstAudioFilter * base,
+ GstRingBufferSpec * format)
+{
+ GstAudioFXBaseFIRFilter *self = GST_AUDIO_FX_BASE_FIR_FILTER (base);
+ gboolean ret = TRUE;
+
+ if (self->residue) {
+ gst_audio_fx_base_fir_filter_push_residue (self);
+ g_free (self->residue);
+ self->residue = NULL;
+ self->residue_length = 0;
+ self->next_ts = GST_CLOCK_TIME_NONE;
+ self->next_off = GST_BUFFER_OFFSET_NONE;
+ }
+
+ if (format->width == 32)
+ self->process = (GstAudioFXBaseFIRFilterProcessFunc) process_32;
+ else if (format->width == 64)
+ self->process = (GstAudioFXBaseFIRFilterProcessFunc) process_64;
+ else
+ ret = FALSE;
+
+ return TRUE;
+}
+
+/* GstBaseTransform vmethod implementations */
+
+static GstFlowReturn
+gst_audio_fx_base_fir_filter_transform (GstBaseTransform * base,
+ GstBuffer * inbuf, GstBuffer * outbuf)
+{
+ GstAudioFXBaseFIRFilter *self = GST_AUDIO_FX_BASE_FIR_FILTER (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 = 0;
+
+ timestamp = GST_BUFFER_TIMESTAMP (outbuf);
+ if (!GST_CLOCK_TIME_IS_VALID (timestamp)) {
+ GST_ERROR_OBJECT (self, "Invalid timestamp");
+ return GST_FLOW_ERROR;
+ }
+
+ gst_object_sync_values (G_OBJECT (self), timestamp);
+
+ g_return_val_if_fail (self->kernel != NULL, GST_FLOW_ERROR);
+ g_return_val_if_fail (channels != 0, GST_FLOW_ERROR);
+
+ if (!self->residue)
+ self->residue = g_new0 (gdouble, self->kernel_length * channels);
+
+ /* Reset the residue if already existing on discont buffers */
+ if (GST_BUFFER_IS_DISCONT (inbuf) || (GST_CLOCK_TIME_IS_VALID (self->next_ts)
+ && timestamp - gst_util_uint64_scale (MIN (self->latency,
+ self->residue_length / channels), GST_SECOND,
+ rate) - self->next_ts > 5 * GST_MSECOND)) {
+ GST_DEBUG_OBJECT (self, "Discontinuity detected - flushing");
+ if (GST_CLOCK_TIME_IS_VALID (self->next_ts))
+ gst_audio_fx_base_fir_filter_push_residue (self);
+ self->residue_length = 0;
+ self->next_ts = timestamp;
+ self->next_off = GST_BUFFER_OFFSET (inbuf);
+ } else if (!GST_CLOCK_TIME_IS_VALID (self->next_ts)) {
+ self->next_ts = timestamp;
+ self->next_off = GST_BUFFER_OFFSET (inbuf);
+ }
+
+ /* Calculate the number of samples we can push out now without outputting
+ * latency zeros in the beginning */
+ diff = self->latency * 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) {
+ return GST_BASE_TRANSFORM_FLOW_DROPPED;
+ }
+
+ GST_BUFFER_TIMESTAMP (outbuf) = self->next_ts;
+ GST_BUFFER_DURATION (outbuf) =
+ gst_util_uint64_scale (output_samples / channels, GST_SECOND, rate);
+ GST_BUFFER_OFFSET (outbuf) = self->next_off;
+ if (GST_BUFFER_OFFSET_IS_VALID (outbuf))
+ GST_BUFFER_OFFSET_END (outbuf) = self->next_off + output_samples / channels;
+ else
+ GST_BUFFER_OFFSET_END (outbuf) = GST_BUFFER_OFFSET_NONE;
+
+ if (output_samples < input_samples) {
+ GST_BUFFER_DATA (outbuf) +=
+ diff * (GST_AUDIO_FILTER (self)->format.width / 8);
+ GST_BUFFER_SIZE (outbuf) -=
+ diff * (GST_AUDIO_FILTER (self)->format.width / 8);
+ }
+
+ self->next_ts += GST_BUFFER_DURATION (outbuf);
+ self->next_off = GST_BUFFER_OFFSET_END (outbuf);
+
+ 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);
+
+ return GST_FLOW_OK;
+}
+
+static gboolean
+gst_audio_fx_base_fir_filter_start (GstBaseTransform * base)
+{
+ GstAudioFXBaseFIRFilter *self = GST_AUDIO_FX_BASE_FIR_FILTER (base);
+
+ self->residue_length = 0;
+ self->next_ts = GST_CLOCK_TIME_NONE;
+ self->next_off = GST_BUFFER_OFFSET_NONE;
+
+ return TRUE;
+}
+
+static gboolean
+gst_audio_fx_base_fir_filter_stop (GstBaseTransform * base)
+{
+ GstAudioFXBaseFIRFilter *self = GST_AUDIO_FX_BASE_FIR_FILTER (base);
+
+ g_free (self->residue);
+ self->residue = NULL;
+
+ return TRUE;
+}
+
+static gboolean
+gst_audio_fx_base_fir_filter_query (GstPad * pad, GstQuery * query)
+{
+ GstAudioFXBaseFIRFilter *self =
+ 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 (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->next_ts = GST_CLOCK_TIME_NONE;
+ self->next_off = GST_BUFFER_OFFSET_NONE;
+ 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->residue) {
+ gst_audio_fx_base_fir_filter_push_residue (self);
+ self->next_ts = GST_CLOCK_TIME_NONE;
+ self->next_off = GST_BUFFER_OFFSET_NONE;
+ self->residue_length = 0;
+ }
+
+ g_free (self->kernel);
+ g_free (self->residue);
+
+ self->kernel = kernel;
+ self->kernel_length = kernel_length;
+
+ if (GST_AUDIO_FILTER (self)->format.channels) {
+ self->residue =
+ g_new0 (gdouble,
+ kernel_length * GST_AUDIO_FILTER (self)->format.channels);
+ self->residue_length = 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);
+}
--- /dev/null
+/* -*- c-basic-offset: 2 -*-
+ *
+ * GStreamer
+ * Copyright (C) 1999-2001 Erik Walthinsen <omega@cse.ogi.edu>
+ * 2006 Dreamlab Technologies Ltd. <mathis.hofer@dreamlab.net>
+ * 2009 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
+ * Boston, MA 02111-1307, USA.
+ *
+ */
+
+#ifndef __GST_AUDIO_FX_BASE_FIR_FILTER_H__
+#define __GST_AUDIO_FX_BASE_FIR_FILTER_H__
+
+#include <gst/gst.h>
+#include <gst/audio/gstaudiofilter.h>
+
+G_BEGIN_DECLS
+
+#define GST_TYPE_AUDIO_FX_BASE_FIR_FILTER \
+ (gst_audio_fx_base_fir_filter_get_type())
+#define GST_AUDIO_FX_BASE_FIR_FILTER(obj) \
+ (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIO_FX_BASE_FIR_FILTER,GstAudioFXBaseFIRFilter))
+#define GST_AUDIO_FX_BASE_FIR_FILTER_CLASS(klass) \
+ (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_AUDIO_FX_BASE_FIR_FILTER,GstAudioFXBaseFIRFilterClass))
+#define GST_IS_AUDIO_FX_BASE_FIR_FILTER(obj) \
+ (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIO_FX_BASE_FIR_FILTER))
+#define GST_IS_AUDIO_FX_BASE_FIR_FILTER_CLASS(klass) \
+ (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_AUDIO_FX_BASE_FIR_FILTER))
+
+typedef struct _GstAudioFXBaseFIRFilter GstAudioFXBaseFIRFilter;
+typedef struct _GstAudioFXBaseFIRFilterClass GstAudioFXBaseFIRFilterClass;
+
+typedef void (*GstAudioFXBaseFIRFilterProcessFunc) (GstAudioFXBaseFIRFilter *, guint8 *, guint8 *, guint);
+
+/**
+ * GstAudioFXBaseFIRFilter:
+ *
+ * Opaque data structure.
+ */
+struct _GstAudioFXBaseFIRFilter {
+ GstAudioFilter element;
+
+ /* < private > */
+ GstAudioFXBaseFIRFilterProcessFunc process;
+
+ gdouble *kernel; /* filter kernel */
+ guint kernel_length; /* length of the filter kernel */
+ gdouble *residue; /* buffer for left-over samples from previous buffer */
+ guint residue_length;
+
+ guint64 latency;
+
+ GstClockTime next_ts;
+ guint64 next_off;
+};
+
+struct _GstAudioFXBaseFIRFilterClass {
+ GstAudioFilterClass parent_class;
+};
+
+GType gst_audio_fx_base_fir_filter_get_type (void);
+void gst_audio_fx_base_fir_filter_set_kernel (GstAudioFXBaseFIRFilter *filter, gdouble *kernel, guint kernel_length, guint64 latency);
+void gst_audio_fx_base_fir_filter_push_residue (GstAudioFXBaseFIRFilter *filter);
+
+G_END_DECLS
+
+#endif /* __GST_AUDIO_FX_BASE_FIR_FILTER_H__ */
" channels = (int) [ 1, MAX ]"
#define DEBUG_INIT(bla) \
- GST_DEBUG_CATEGORY_INIT (gst_audio_fx_base_iir_filter_debug, "audiobaseiirfilter", 0, "Audio IIR Filter Base Class");
+ GST_DEBUG_CATEGORY_INIT (gst_audio_fx_base_iir_filter_debug, "audiofxbaseiirfilter", 0, "Audio IIR Filter Base Class");
GST_BOILERPLATE_FULL (GstAudioFXBaseIIRFilter,
gst_audio_fx_base_iir_filter, GstAudioFilter, GST_TYPE_AUDIO_FILTER,
* GStreamer
* Copyright (C) 1999-2001 Erik Walthinsen <omega@cse.ogi.edu>
* 2006 Dreamlab Technologies Ltd. <mathis.hofer@dreamlab.net>
- * 2007 Sebastian Dröge <slomo@circular-chaos.org>
+ * 2007-2009 Sebastian Dröge <sebastian.droege@collabora.co.uk>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
#include "audiowsincband.h"
-#define GST_CAT_DEFAULT gst_audio_wsincband_debug
+#define GST_CAT_DEFAULT gst_gst_audio_wsincband_debug
GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
-static const GstElementDetails audio_wsincband_details =
-GST_ELEMENT_DETAILS ("Band pass & band reject filter",
- "Filter/Effect/Audio",
- "Band pass and band reject windowed sinc filter",
- "Thomas Vander Stichele <thomas at apestaart dot org>, "
- "Steven W. Smith, "
- "Dreamlab Technologies Ltd. <mathis.hofer@dreamlab.net>, "
- "Sebastian Dröge <slomo@circular-chaos.org>");
-
-/* Filter signals and args */
-enum
-{
- /* FILL ME */
- LAST_SIGNAL
-};
-
enum
{
PROP_0,
MODE_BAND_REJECT
};
-#define GST_TYPE_AUDIO_WSINC_BAND_MODE (gst_audio_wsincband_mode_get_type ())
+#define GST_TYPE_AUDIO_WSINC_BAND_MODE (gst_gst_audio_wsincband_mode_get_type ())
static GType
-gst_audio_wsincband_mode_get_type (void)
+gst_gst_audio_wsincband_mode_get_type (void)
{
static GType gtype = 0;
WINDOW_BLACKMAN
};
-#define GST_TYPE_AUDIO_WSINC_BAND_WINDOW (gst_audio_wsincband_window_get_type ())
+#define GST_TYPE_AUDIO_WSINC_BAND_WINDOW (gst_gst_audio_wsincband_window_get_type ())
static GType
-gst_audio_wsincband_window_get_type (void)
+gst_gst_audio_wsincband_window_get_type (void)
{
static GType gtype = 0;
return gtype;
}
-#define ALLOWED_CAPS \
- "audio/x-raw-float, " \
- " width = (int) { 32, 64 }, " \
- " endianness = (int) BYTE_ORDER, " \
- " rate = (int) [ 1, MAX ], " \
- " channels = (int) [ 1, MAX ] "
-
#define DEBUG_INIT(bla) \
- GST_DEBUG_CATEGORY_INIT (gst_audio_wsincband_debug, "audiowsincband", 0, \
+ GST_DEBUG_CATEGORY_INIT (gst_gst_audio_wsincband_debug, "audiowsincband", 0, \
"Band-pass and Band-reject Windowed sinc filter plugin");
-GST_BOILERPLATE_FULL (GstAudioWSincBand, audio_wsincband, GstAudioFilter,
- GST_TYPE_AUDIO_FILTER, DEBUG_INIT);
+GST_BOILERPLATE_FULL (GstAudioWSincBand, gst_audio_wsincband, GstAudioFilter,
+ GST_TYPE_AUDIO_FX_BASE_FIR_FILTER, DEBUG_INIT);
-static void audio_wsincband_set_property (GObject * object, guint prop_id,
+static void gst_audio_wsincband_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
-static void audio_wsincband_get_property (GObject * object, guint prop_id,
+static void gst_audio_wsincband_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
-static GstFlowReturn audio_wsincband_transform (GstBaseTransform * base,
- GstBuffer * inbuf, GstBuffer * outbuf);
-static gboolean audio_wsincband_start (GstBaseTransform * base);
-static gboolean audio_wsincband_event (GstBaseTransform * base,
- GstEvent * event);
-
-static gboolean audio_wsincband_setup (GstAudioFilter * base,
+static gboolean gst_audio_wsincband_setup (GstAudioFilter * base,
GstRingBufferSpec * format);
-static gboolean audio_wsincband_query (GstPad * pad, GstQuery * query);
-static const GstQueryType *audio_wsincband_query_type (GstPad * pad);
-
/* Element class */
-
static void
-audio_wsincband_dispose (GObject * object)
-{
- GstAudioWSincBand *self = GST_AUDIO_WSINC_BAND (object);
-
- if (self->residue) {
- g_free (self->residue);
- self->residue = NULL;
- }
-
- if (self->kernel) {
- g_free (self->kernel);
- self->kernel = NULL;
- }
-
- G_OBJECT_CLASS (parent_class)->dispose (object);
-}
-
-static void
-audio_wsincband_base_init (gpointer g_class)
+gst_audio_wsincband_base_init (gpointer g_class)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
- GstCaps *caps;
- gst_element_class_set_details (element_class, &audio_wsincband_details);
-
- caps = gst_caps_from_string (ALLOWED_CAPS);
- gst_audio_filter_class_add_pad_templates (GST_AUDIO_FILTER_CLASS (g_class),
- caps);
- gst_caps_unref (caps);
+ gst_element_class_set_details_simple (element_class,
+ "Band pass & band reject filter", "Filter/Effect/Audio",
+ "Band pass and band reject windowed sinc filter",
+ "Thomas Vander Stichele <thomas at apestaart dot org>, "
+ "Steven W. Smith, "
+ "Dreamlab Technologies Ltd. <mathis.hofer@dreamlab.net>, "
+ "Sebastian Dröge <sebastian.droege@collabora.co.uk>");
}
static void
-audio_wsincband_class_init (GstAudioWSincBandClass * klass)
+gst_audio_wsincband_class_init (GstAudioWSincBandClass * klass)
{
- GObjectClass *gobject_class;
- GstBaseTransformClass *trans_class;
- GstAudioFilterClass *filter_class;
-
- gobject_class = (GObjectClass *) klass;
- trans_class = (GstBaseTransformClass *) klass;
- filter_class = (GstAudioFilterClass *) klass;
+ GObjectClass *gobject_class = (GObjectClass *) klass;
+ GstAudioFilterClass *filter_class = (GstAudioFilterClass *) klass;
- gobject_class->set_property = audio_wsincband_set_property;
- gobject_class->get_property = audio_wsincband_get_property;
- gobject_class->dispose = audio_wsincband_dispose;
+ gobject_class->set_property = gst_audio_wsincband_set_property;
+ gobject_class->get_property = gst_audio_wsincband_get_property;
/* FIXME: Don't use the complete possible range but restrict the upper boundary
* so automatically generated UIs can use a slider */
g_object_class_install_property (gobject_class, PROP_LOWER_FREQUENCY,
g_param_spec_float ("lower-frequency", "Lower Frequency",
- "Cut-off lower frequency (Hz)", 0.0, 100000.0, 0, G_PARAM_READWRITE));
+ "Cut-off lower frequency (Hz)", 0.0, 100000.0, 0,
+ G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_UPPER_FREQUENCY,
g_param_spec_float ("upper-frequency", "Upper Frequency",
- "Cut-off upper frequency (Hz)", 0.0, 100000.0, 0, G_PARAM_READWRITE));
+ "Cut-off upper frequency (Hz)", 0.0, 100000.0, 0,
+ G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_LENGTH,
g_param_spec_int ("length", "Length",
- "Filter kernel length, will be rounded to the next odd number",
- 3, 50000, 101, G_PARAM_READWRITE));
+ "Filter kernel length, will be rounded to the next odd number", 3,
+ 50000, 101,
+ G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_MODE,
g_param_spec_enum ("mode", "Mode",
"Band pass or band reject mode", GST_TYPE_AUDIO_WSINC_BAND_MODE,
- MODE_BAND_PASS, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
+ MODE_BAND_PASS,
+ G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_WINDOW,
g_param_spec_enum ("window", "Window",
"Window function to use", GST_TYPE_AUDIO_WSINC_BAND_WINDOW,
- WINDOW_HAMMING, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
+ WINDOW_HAMMING,
+ G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE | G_PARAM_STATIC_STRINGS));
- trans_class->transform = GST_DEBUG_FUNCPTR (audio_wsincband_transform);
- trans_class->start = GST_DEBUG_FUNCPTR (audio_wsincband_start);
- trans_class->event = GST_DEBUG_FUNCPTR (audio_wsincband_event);
- filter_class->setup = GST_DEBUG_FUNCPTR (audio_wsincband_setup);
+ filter_class->setup = GST_DEBUG_FUNCPTR (gst_audio_wsincband_setup);
}
static void
-audio_wsincband_init (GstAudioWSincBand * self,
+gst_audio_wsincband_init (GstAudioWSincBand * self,
GstAudioWSincBandClass * g_class)
{
self->kernel_length = 101;
- self->latency = 50;
self->lower_frequency = 0.0;
self->upper_frequency = 0.0;
self->mode = MODE_BAND_PASS;
self->window = WINDOW_HAMMING;
- self->kernel = NULL;
- self->have_kernel = FALSE;
- self->residue = NULL;
-
- self->residue_length = 0;
- self->next_ts = GST_CLOCK_TIME_NONE;
- self->next_off = GST_BUFFER_OFFSET_NONE;
-
- gst_pad_set_query_function (GST_BASE_TRANSFORM (self)->srcpad,
- audio_wsincband_query);
- gst_pad_set_query_type_function (GST_BASE_TRANSFORM (self)->srcpad,
- audio_wsincband_query_type);
-}
-
-#define DEFINE_PROCESS_FUNC(width,ctype) \
-static void \
-process_##width (GstAudioWSincBand * self, g##ctype * src, g##ctype * dst, guint input_samples) \
-{ \
- gint kernel_length = self->kernel_length; \
- gint i, j, k, l; \
- gint channels = GST_AUDIO_FILTER (self)->format.channels; \
- gint res_start; \
- \
- /* convolution */ \
- for (i = 0; i < input_samples; i++) { \
- dst[i] = 0.0; \
- k = i % channels; \
- l = i / channels; \
- for (j = 0; j < kernel_length; j++) \
- if (l < j) \
- dst[i] += \
- self->residue[(kernel_length + l - j) * channels + \
- k] * self->kernel[j]; \
- else \
- dst[i] += src[(l - j) * channels + k] * self->kernel[j]; \
- } \
- \
- /* copy the tail of the current input buffer to the residue, while \
- * keeping parts of the residue if the input buffer is smaller than \
- * the kernel length */ \
- if (input_samples < kernel_length * channels) \
- res_start = kernel_length * channels - input_samples; \
- else \
- res_start = 0; \
- \
- for (i = 0; i < res_start; i++) \
- self->residue[i] = self->residue[i + input_samples]; \
- for (i = res_start; i < kernel_length * channels; i++) \
- self->residue[i] = src[input_samples - kernel_length * channels + i]; \
- \
- self->residue_length += kernel_length * channels - res_start; \
- if (self->residue_length > kernel_length * channels) \
- self->residue_length = kernel_length * channels; \
}
-DEFINE_PROCESS_FUNC (32, float);
-DEFINE_PROCESS_FUNC (64, double);
-
-#undef DEFINE_PROCESS_FUNC
-
static void
-audio_wsincband_build_kernel (GstAudioWSincBand * self)
+gst_audio_wsincband_build_kernel (GstAudioWSincBand * self)
{
gint i = 0;
gdouble sum = 0.0;
gint len = 0;
gdouble *kernel_lp, *kernel_hp;
gdouble w;
+ gdouble *kernel;
len = self->kernel_length;
self->upper_frequency = tmp;
}
- GST_DEBUG ("audio_wsincband: initializing filter kernel of length %d "
+ GST_DEBUG ("gst_audio_wsincband: initializing filter kernel of length %d "
"with lower frequency %.2lf Hz "
", upper frequency %.2lf Hz for mode %s",
len, self->lower_frequency, self->upper_frequency,
kernel_hp[len / 2] += 1;
/* combine the two kernels */
- if (self->kernel)
- g_free (self->kernel);
- self->kernel = g_new (gdouble, len);
+ kernel = g_new (gdouble, len);
for (i = 0; i < len; ++i)
- self->kernel[i] = kernel_lp[i] + kernel_hp[i];
+ kernel[i] = kernel_lp[i] + kernel_hp[i];
/* free the helper kernels */
g_free (kernel_lp);
* if specified */
if (self->mode == MODE_BAND_PASS) {
for (i = 0; i < len; ++i)
- self->kernel[i] = -self->kernel[i];
- self->kernel[len / 2] += 1;
- }
-
- /* set up the residue memory space */
- if (!self->residue) {
- self->residue =
- g_new0 (gdouble, len * GST_AUDIO_FILTER (self)->format.channels);
- self->residue_length = 0;
- }
-
- self->have_kernel = TRUE;
-}
-
-static void
-audio_wsincband_push_residue (GstAudioWSincBand * self)
-{
- GstBuffer *outbuf;
- GstFlowReturn res;
- gint rate = GST_AUDIO_FILTER (self)->format.rate;
- gint channels = GST_AUDIO_FILTER (self)->format.channels;
- gint outsize, outsamples;
- gint diffsize, diffsamples;
- guint8 *in, *out;
-
- /* Calculate the number of samples and their memory size that
- * should be pushed from the residue */
- outsamples = MIN (self->latency, self->residue_length / channels);
- outsize = outsamples * channels * (GST_AUDIO_FILTER (self)->format.width / 8);
- if (outsize == 0)
- return;
-
- /* Process the difference between latency and residue_length samples
- * to start at the actual data instead of starting at the zeros before
- * when we only got one buffer smaller than latency */
- diffsamples = self->latency - self->residue_length / channels;
- diffsize =
- diffsamples * channels * (GST_AUDIO_FILTER (self)->format.width / 8);
- if (diffsize > 0) {
- in = g_new0 (guint8, diffsize);
- out = g_new0 (guint8, diffsize);
- self->process (self, in, out, diffsamples * channels);
- g_free (in);
- g_free (out);
- }
-
- res = gst_pad_alloc_buffer (GST_BASE_TRANSFORM (self)->srcpad,
- GST_BUFFER_OFFSET_NONE, outsize,
- GST_PAD_CAPS (GST_BASE_TRANSFORM (self)->srcpad), &outbuf);
-
- if (G_UNLIKELY (res != GST_FLOW_OK)) {
- GST_WARNING_OBJECT (self, "failed allocating buffer of %d bytes", outsize);
- return;
- }
-
- /* Convolve the residue with zeros to get the actual remaining data */
- in = g_new0 (guint8, outsize);
- self->process (self, in, GST_BUFFER_DATA (outbuf), outsamples * channels);
- g_free (in);
-
- /* Set timestamp, offset, etc from the values we
- * saved when processing the regular buffers */
- if (GST_CLOCK_TIME_IS_VALID (self->next_ts))
- GST_BUFFER_TIMESTAMP (outbuf) = self->next_ts;
- else
- GST_BUFFER_TIMESTAMP (outbuf) = 0;
- GST_BUFFER_DURATION (outbuf) =
- gst_util_uint64_scale (outsamples, GST_SECOND, rate);
- self->next_ts += gst_util_uint64_scale (outsamples, GST_SECOND, rate);
-
- 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");
+ kernel[i] = -kernel[i];
+ kernel[len / 2] += 1;
}
+ gst_audio_fx_base_fir_filter_set_kernel (GST_AUDIO_FX_BASE_FIR_FILTER (self),
+ kernel, self->kernel_length, (len - 1) / 2);
}
/* GstAudioFilter vmethod implementations */
/* get notified of caps and plug in the correct process function */
static gboolean
-audio_wsincband_setup (GstAudioFilter * base, GstRingBufferSpec * format)
-{
- GstAudioWSincBand *self = GST_AUDIO_WSINC_BAND (base);
-
- gboolean ret = TRUE;
-
- if (format->width == 32)
- self->process = (GstAudioWSincBandProcessFunc) process_32;
- else if (format->width == 64)
- self->process = (GstAudioWSincBandProcessFunc) process_64;
- else
- ret = FALSE;
-
- self->have_kernel = FALSE;
-
- return TRUE;
-}
-
-/* GstBaseTransform vmethod implementations */
-
-static GstFlowReturn
-audio_wsincband_transform (GstBaseTransform * base, GstBuffer * inbuf,
- GstBuffer * outbuf)
-{
- GstAudioWSincBand *self = GST_AUDIO_WSINC_BAND (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_wsincband_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_wsincband_start (GstBaseTransform * base)
-{
- GstAudioWSincBand *self = GST_AUDIO_WSINC_BAND (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_wsincband_query (GstPad * pad, GstQuery * query)
-{
- GstAudioWSincBand *self = GST_AUDIO_WSINC_BAND (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_wsincband_query_type (GstPad * pad)
-{
- static const GstQueryType types[] = {
- GST_QUERY_LATENCY,
- 0
- };
-
- return types;
-}
-
-static gboolean
-audio_wsincband_event (GstBaseTransform * base, GstEvent * event)
+gst_audio_wsincband_setup (GstAudioFilter * base, GstRingBufferSpec * format)
{
GstAudioWSincBand *self = GST_AUDIO_WSINC_BAND (base);
- switch (GST_EVENT_TYPE (event)) {
- case GST_EVENT_EOS:
- audio_wsincband_push_residue (self);
- break;
- default:
- break;
- }
+ gst_audio_wsincband_build_kernel (self);
- return GST_BASE_TRANSFORM_CLASS (parent_class)->event (base, event);
+ return GST_AUDIO_FILTER_CLASS (parent_class)->setup (base, format);
}
static void
-audio_wsincband_set_property (GObject * object, guint prop_id,
+gst_audio_wsincband_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstAudioWSincBand *self = GST_AUDIO_WSINC_BAND (object);
case PROP_LENGTH:{
gint val;
- GST_BASE_TRANSFORM_LOCK (self);
+ GST_OBJECT_LOCK (self);
val = g_value_get_int (value);
if (val % 2 == 0)
val++;
if (val != self->kernel_length) {
- if (self->residue) {
- audio_wsincband_push_residue (self);
- g_free (self->residue);
- self->residue = NULL;
- }
+ gst_audio_fx_base_fir_filter_push_residue (GST_AUDIO_FX_BASE_FIR_FILTER
+ (self));
self->kernel_length = val;
- self->latency = val / 2;
- audio_wsincband_build_kernel (self);
- gst_element_post_message (GST_ELEMENT (self),
- gst_message_new_latency (GST_OBJECT (self)));
+ gst_audio_wsincband_build_kernel (self);
}
- GST_BASE_TRANSFORM_UNLOCK (self);
+ GST_OBJECT_UNLOCK (self);
break;
}
case PROP_LOWER_FREQUENCY:
- GST_BASE_TRANSFORM_LOCK (self);
+ GST_OBJECT_LOCK (self);
self->lower_frequency = g_value_get_float (value);
- audio_wsincband_build_kernel (self);
- GST_BASE_TRANSFORM_UNLOCK (self);
+ gst_audio_wsincband_build_kernel (self);
+ GST_OBJECT_UNLOCK (self);
break;
case PROP_UPPER_FREQUENCY:
- GST_BASE_TRANSFORM_LOCK (self);
+ GST_OBJECT_LOCK (self);
self->upper_frequency = g_value_get_float (value);
- audio_wsincband_build_kernel (self);
- GST_BASE_TRANSFORM_UNLOCK (self);
+ gst_audio_wsincband_build_kernel (self);
+ GST_OBJECT_UNLOCK (self);
break;
case PROP_MODE:
- GST_BASE_TRANSFORM_LOCK (self);
+ GST_OBJECT_LOCK (self);
self->mode = g_value_get_enum (value);
- audio_wsincband_build_kernel (self);
- GST_BASE_TRANSFORM_UNLOCK (self);
+ gst_audio_wsincband_build_kernel (self);
+ GST_OBJECT_UNLOCK (self);
break;
case PROP_WINDOW:
- GST_BASE_TRANSFORM_LOCK (self);
+ GST_OBJECT_LOCK (self);
self->window = g_value_get_enum (value);
- audio_wsincband_build_kernel (self);
- GST_BASE_TRANSFORM_UNLOCK (self);
+ gst_audio_wsincband_build_kernel (self);
+ GST_OBJECT_UNLOCK (self);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
}
static void
-audio_wsincband_get_property (GObject * object, guint prop_id, GValue * value,
- GParamSpec * pspec)
+gst_audio_wsincband_get_property (GObject * object, guint prop_id,
+ GValue * value, GParamSpec * pspec)
{
GstAudioWSincBand *self = GST_AUDIO_WSINC_BAND (object);
* GStreamer
* Copyright (C) 1999-2001 Erik Walthinsen <omega@cse.ogi.edu>
* 2006 Dreamlab Technologies Ltd. <mathis.hofer@dreamlab.net>
+ * 2007-2009 Sebastian Dröge <sebastian.droege@collabora.co.uk>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
#include <gst/gst.h>
#include <gst/audio/gstaudiofilter.h>
+#include "audiofxbasefirfilter.h"
+
G_BEGIN_DECLS
#define GST_TYPE_AUDIO_WSINC_BAND \
- (audio_wsincband_get_type())
+ (gst_audio_wsincband_get_type())
#define GST_AUDIO_WSINC_BAND(obj) \
(G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIO_WSINC_BAND,GstAudioWSincBand))
#define GST_AUDIO_WSINC_BAND_CLASS(klass) \
typedef struct _GstAudioWSincBand GstAudioWSincBand;
typedef struct _GstAudioWSincBandClass GstAudioWSincBandClass;
-typedef void (*GstAudioWSincBandProcessFunc) (GstAudioWSincBand *, guint8 *, guint8 *, guint);
-
/**
* GstAudioWSincBand:
*
* Opaque data structure.
*/
struct _GstAudioWSincBand {
- GstAudioFilter element;
+ GstAudioFXBaseFIRFilter parent;
/* < private > */
- GstAudioWSincBandProcessFunc process;
-
gint mode;
gint window;
gfloat lower_frequency, upper_frequency;
gint kernel_length; /* length of the filter kernel */
-
- gdouble *residue; /* buffer for left-over samples from previous buffer */
- gdouble *kernel;
- gboolean have_kernel;
- gint residue_length;
- guint64 latency;
- GstClockTime next_ts;
- guint64 next_off;
};
struct _GstAudioWSincBandClass {
- GstAudioFilterClass parent_class;
+ GstAudioFilterClass parent;
};
-GType audio_wsincband_get_type (void);
+GType gst_audio_wsincband_get_type (void);
G_END_DECLS
* GStreamer
* Copyright (C) 1999-2001 Erik Walthinsen <omega@cse.ogi.edu>
* 2006 Dreamlab Technologies Ltd. <mathis.hofer@dreamlab.net>
- * 2007 Sebastian Dröge <slomo@circular-chaos.org>
+ * 2007-2009 Sebastian Dröge <sebastian.droege@collabora.co.uk>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
#include "audiowsinclimit.h"
-#define GST_CAT_DEFAULT audio_wsinclimit_debug
+#define GST_CAT_DEFAULT gst_audio_wsinclimit_debug
GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
-static const GstElementDetails audio_wsinclimit_details =
-GST_ELEMENT_DETAILS ("Low pass & high pass filter",
- "Filter/Effect/Audio",
- "Low pass and high pass windowed sinc filter",
- "Thomas Vander Stichele <thomas at apestaart dot org>, "
- "Steven W. Smith, "
- "Dreamlab Technologies Ltd. <mathis.hofer@dreamlab.net>, "
- "Sebastian Dröge <slomo@circular-chaos.org>");
-
-/* Filter signals and args */
-enum
-{
- /* FILL ME */
- LAST_SIGNAL
-};
-
enum
{
PROP_0,
MODE_HIGH_PASS
};
-#define GST_TYPE_AUDIO_WSINC_LIMIT_MODE (audio_wsinclimit_mode_get_type ())
+#define GST_TYPE_AUDIO_WSINC_LIMIT_MODE (gst_audio_wsinclimit_mode_get_type ())
static GType
-audio_wsinclimit_mode_get_type (void)
+gst_audio_wsinclimit_mode_get_type (void)
{
static GType gtype = 0;
WINDOW_BLACKMAN
};
-#define GST_TYPE_AUDIO_WSINC_LIMIT_WINDOW (audio_wsinclimit_window_get_type ())
+#define GST_TYPE_AUDIO_WSINC_LIMIT_WINDOW (gst_audio_wsinclimit_window_get_type ())
static GType
-audio_wsinclimit_window_get_type (void)
+gst_audio_wsinclimit_window_get_type (void)
{
static GType gtype = 0;
return gtype;
}
-#define ALLOWED_CAPS \
- "audio/x-raw-float, " \
- " width = (int) { 32, 64 }, " \
- " endianness = (int) BYTE_ORDER, " \
- " rate = (int) [ 1, MAX ], " \
- " channels = (int) [ 1, MAX ]"
-
#define DEBUG_INIT(bla) \
- GST_DEBUG_CATEGORY_INIT (audio_wsinclimit_debug, "audiowsinclimit", 0, \
+ GST_DEBUG_CATEGORY_INIT (gst_audio_wsinclimit_debug, "audiowsinclimit", 0, \
"Low-pass and High-pass Windowed sinc filter plugin");
-GST_BOILERPLATE_FULL (GstAudioWSincLimit, audio_wsinclimit, GstAudioFilter,
- GST_TYPE_AUDIO_FILTER, DEBUG_INIT);
+GST_BOILERPLATE_FULL (GstAudioWSincLimit, gst_audio_wsinclimit, GstAudioFilter,
+ GST_TYPE_AUDIO_FX_BASE_FIR_FILTER, DEBUG_INIT);
-static void audio_wsinclimit_set_property (GObject * object, guint prop_id,
+static void gst_audio_wsinclimit_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
-static void audio_wsinclimit_get_property (GObject * object, guint prop_id,
+static void gst_audio_wsinclimit_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
-static GstFlowReturn audio_wsinclimit_transform (GstBaseTransform * base,
- GstBuffer * inbuf, GstBuffer * outbuf);
-static gboolean audio_wsinclimit_start (GstBaseTransform * base);
-static gboolean audio_wsinclimit_event (GstBaseTransform * base,
- GstEvent * event);
-static gboolean audio_wsinclimit_setup (GstAudioFilter * base,
+static gboolean gst_audio_wsinclimit_setup (GstAudioFilter * base,
GstRingBufferSpec * format);
-static gboolean audio_wsinclimit_query (GstPad * pad, GstQuery * query);
-static const GstQueryType *audio_wsinclimit_query_type (GstPad * pad);
-
/* Element class */
static void
-audio_wsinclimit_dispose (GObject * object)
-{
- GstAudioWSincLimit *self = GST_AUDIO_WSINC_LIMIT (object);
-
- if (self->residue) {
- g_free (self->residue);
- self->residue = NULL;
- }
-
- if (self->kernel) {
- g_free (self->kernel);
- self->kernel = NULL;
- }
-
- G_OBJECT_CLASS (parent_class)->dispose (object);
-}
-
-static void
-audio_wsinclimit_base_init (gpointer g_class)
+gst_audio_wsinclimit_base_init (gpointer g_class)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
- GstCaps *caps;
-
- gst_element_class_set_details (element_class, &audio_wsinclimit_details);
- caps = gst_caps_from_string (ALLOWED_CAPS);
- gst_audio_filter_class_add_pad_templates (GST_AUDIO_FILTER_CLASS (g_class),
- caps);
- gst_caps_unref (caps);
+ gst_element_class_set_details_simple (element_class,
+ "Low pass & high pass filter", "Filter/Effect/Audio",
+ "Low pass and high pass windowed sinc filter",
+ "Thomas Vander Stichele <thomas at apestaart dot org>, "
+ "Steven W. Smith, "
+ "Dreamlab Technologies Ltd. <mathis.hofer@dreamlab.net>, "
+ "Sebastian Dröge <sebastian.droege@collabora.co.uk>");
}
static void
-audio_wsinclimit_class_init (GstAudioWSincLimitClass * klass)
+gst_audio_wsinclimit_class_init (GstAudioWSincLimitClass * klass)
{
- GObjectClass *gobject_class;
- GstBaseTransformClass *trans_class;
- GstAudioFilterClass *filter_class;
-
- gobject_class = (GObjectClass *) klass;
- trans_class = (GstBaseTransformClass *) klass;
- filter_class = (GstAudioFilterClass *) klass;
-
- gobject_class->set_property = audio_wsinclimit_set_property;
- gobject_class->get_property = audio_wsinclimit_get_property;
- gobject_class->dispose = audio_wsinclimit_dispose;
+ GObjectClass *gobject_class = (GObjectClass *) klass;
+ GstAudioFilterClass *filter_class = (GstAudioFilterClass *) klass;
+ gobject_class->set_property = gst_audio_wsinclimit_set_property;
+ gobject_class->get_property = gst_audio_wsinclimit_get_property;
/* FIXME: Don't use the complete possible range but restrict the upper boundary
* so automatically generated UIs can use a slider */
g_object_class_install_property (gobject_class, PROP_FREQUENCY,
g_param_spec_float ("cutoff", "Cutoff",
"Cut-off Frequency (Hz)", 0.0, 100000.0, 0.0,
- G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
+ G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_LENGTH,
g_param_spec_int ("length", "Length",
"Filter kernel length, will be rounded to the next odd number",
- 3, 50000, 101, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
+ 3, 50000, 101,
+ G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_MODE,
g_param_spec_enum ("mode", "Mode",
"Low pass or high pass mode", GST_TYPE_AUDIO_WSINC_LIMIT_MODE,
- MODE_LOW_PASS, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
+ MODE_LOW_PASS,
+ G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_WINDOW,
g_param_spec_enum ("window", "Window",
"Window function to use", GST_TYPE_AUDIO_WSINC_LIMIT_WINDOW,
- WINDOW_HAMMING, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
+ WINDOW_HAMMING,
+ G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE | G_PARAM_STATIC_STRINGS));
- trans_class->transform = GST_DEBUG_FUNCPTR (audio_wsinclimit_transform);
- trans_class->start = GST_DEBUG_FUNCPTR (audio_wsinclimit_start);
- trans_class->event = GST_DEBUG_FUNCPTR (audio_wsinclimit_event);
- filter_class->setup = GST_DEBUG_FUNCPTR (audio_wsinclimit_setup);
+ filter_class->setup = GST_DEBUG_FUNCPTR (gst_audio_wsinclimit_setup);
}
static void
-audio_wsinclimit_init (GstAudioWSincLimit * self,
+gst_audio_wsinclimit_init (GstAudioWSincLimit * self,
GstAudioWSincLimitClass * g_class)
{
self->mode = MODE_LOW_PASS;
self->window = WINDOW_HAMMING;
self->kernel_length = 101;
- self->latency = 50;
self->cutoff = 0.0;
- self->kernel = NULL;
- self->residue = NULL;
-
- self->have_kernel = FALSE;
- self->residue_length = 0;
- self->next_ts = GST_CLOCK_TIME_NONE;
- self->next_off = GST_BUFFER_OFFSET_NONE;
-
- gst_pad_set_query_function (GST_BASE_TRANSFORM (self)->srcpad,
- audio_wsinclimit_query);
- gst_pad_set_query_type_function (GST_BASE_TRANSFORM (self)->srcpad,
- audio_wsinclimit_query_type);
-}
-
-#define DEFINE_PROCESS_FUNC(width,ctype) \
-static void \
-process_##width (GstAudioWSincLimit * self, g##ctype * src, g##ctype * dst, guint input_samples) \
-{ \
- gint kernel_length = self->kernel_length; \
- gint i, j, k, l; \
- gint channels = GST_AUDIO_FILTER (self)->format.channels; \
- gint res_start; \
- \
- /* convolution */ \
- for (i = 0; i < input_samples; i++) { \
- dst[i] = 0.0; \
- k = i % channels; \
- l = i / channels; \
- for (j = 0; j < kernel_length; j++) \
- if (l < j) \
- dst[i] += \
- self->residue[(kernel_length + l - j) * channels + \
- k] * self->kernel[j]; \
- else \
- dst[i] += src[(l - j) * channels + k] * self->kernel[j]; \
- } \
- \
- /* copy the tail of the current input buffer to the residue, while \
- * keeping parts of the residue if the input buffer is smaller than \
- * the kernel length */ \
- if (input_samples < kernel_length * channels) \
- res_start = kernel_length * channels - input_samples; \
- else \
- res_start = 0; \
- \
- for (i = 0; i < res_start; i++) \
- self->residue[i] = self->residue[i + input_samples]; \
- for (i = res_start; i < kernel_length * channels; i++) \
- self->residue[i] = src[input_samples - kernel_length * channels + i]; \
- \
- self->residue_length += kernel_length * channels - res_start; \
- if (self->residue_length > kernel_length * channels) \
- self->residue_length = kernel_length * channels; \
}
-DEFINE_PROCESS_FUNC (32, float);
-DEFINE_PROCESS_FUNC (64, double);
-
-#undef DEFINE_PROCESS_FUNC
-
static void
-audio_wsinclimit_build_kernel (GstAudioWSincLimit * self)
+gst_audio_wsinclimit_build_kernel (GstAudioWSincLimit * self)
{
gint i = 0;
gdouble sum = 0.0;
gint len = 0;
gdouble w;
+ gdouble *kernel = NULL;
len = self->kernel_length;
self->cutoff =
CLAMP (self->cutoff, 0.0, GST_AUDIO_FILTER (self)->format.rate / 2);
- GST_DEBUG ("audio_wsinclimit_: initializing filter kernel of length %d "
+ GST_DEBUG ("gst_audio_wsinclimit_: initializing filter kernel of length %d "
"with cutoff %.2lf Hz "
"for mode %s",
len, self->cutoff,
/* fill the kernel */
w = 2 * M_PI * (self->cutoff / GST_AUDIO_FILTER (self)->format.rate);
- if (self->kernel)
- g_free (self->kernel);
- self->kernel = g_new (gdouble, len);
+ kernel = g_new (gdouble, len);
for (i = 0; i < len; ++i) {
if (i == len / 2)
- self->kernel[i] = w;
+ kernel[i] = w;
else
- self->kernel[i] = sin (w * (i - len / 2)) / (i - len / 2);
+ kernel[i] = sin (w * (i - len / 2)) / (i - len / 2);
/* windowing */
if (self->window == WINDOW_HAMMING)
- self->kernel[i] *= (0.54 - 0.46 * cos (2 * M_PI * i / len));
+ kernel[i] *= (0.54 - 0.46 * cos (2 * M_PI * i / len));
else
- self->kernel[i] *=
- (0.42 - 0.5 * cos (2 * M_PI * i / len) +
+ kernel[i] *= (0.42 - 0.5 * cos (2 * M_PI * i / len) +
0.08 * cos (4 * M_PI * i / len));
}
/* normalize for unity gain at DC */
for (i = 0; i < len; ++i)
- sum += self->kernel[i];
+ sum += kernel[i];
for (i = 0; i < len; ++i)
- self->kernel[i] /= sum;
+ kernel[i] /= sum;
/* convert to highpass if specified */
if (self->mode == MODE_HIGH_PASS) {
for (i = 0; i < len; ++i)
- self->kernel[i] = -self->kernel[i];
- self->kernel[len / 2] += 1.0;
- }
-
- /* set up the residue memory space */
- if (!self->residue) {
- self->residue =
- g_new0 (gdouble, len * GST_AUDIO_FILTER (self)->format.channels);
- self->residue_length = 0;
- }
-
- self->have_kernel = TRUE;
-}
-
-static void
-audio_wsinclimit_push_residue (GstAudioWSincLimit * self)
-{
- GstBuffer *outbuf;
- GstFlowReturn res;
- gint rate = GST_AUDIO_FILTER (self)->format.rate;
- gint channels = GST_AUDIO_FILTER (self)->format.channels;
- gint outsize, outsamples;
- gint diffsize, diffsamples;
- guint8 *in, *out;
-
- /* Calculate the number of samples and their memory size that
- * should be pushed from the residue */
- outsamples = MIN (self->latency, self->residue_length / channels);
- outsize = outsamples * channels * (GST_AUDIO_FILTER (self)->format.width / 8);
- if (outsize == 0)
- return;
-
- /* Process the difference between latency and residue_length samples
- * to start at the actual data instead of starting at the zeros before
- * when we only got one buffer smaller than latency */
- diffsamples = self->latency - self->residue_length / channels;
- diffsize =
- diffsamples * channels * (GST_AUDIO_FILTER (self)->format.width / 8);
- if (diffsize > 0) {
- in = g_new0 (guint8, diffsize);
- out = g_new0 (guint8, diffsize);
- self->process (self, in, out, diffsamples * channels);
- g_free (in);
- g_free (out);
- }
-
- res = gst_pad_alloc_buffer (GST_BASE_TRANSFORM (self)->srcpad,
- GST_BUFFER_OFFSET_NONE, outsize,
- GST_PAD_CAPS (GST_BASE_TRANSFORM (self)->srcpad), &outbuf);
-
- if (G_UNLIKELY (res != GST_FLOW_OK)) {
- GST_WARNING_OBJECT (self, "failed allocating buffer of %d bytes", outsize);
- return;
- }
-
- /* Convolve the residue with zeros to get the actual remaining data */
- in = g_new0 (guint8, outsize);
- self->process (self, in, GST_BUFFER_DATA (outbuf), outsamples * channels);
- g_free (in);
-
- /* Set timestamp, offset, etc from the values we
- * saved when processing the regular buffers */
- if (GST_CLOCK_TIME_IS_VALID (self->next_ts))
- GST_BUFFER_TIMESTAMP (outbuf) = self->next_ts;
- else
- GST_BUFFER_TIMESTAMP (outbuf) = 0;
- GST_BUFFER_DURATION (outbuf) =
- gst_util_uint64_scale (outsamples, GST_SECOND, rate);
- self->next_ts += gst_util_uint64_scale (outsamples, GST_SECOND, rate);
-
- 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");
+ kernel[i] = -kernel[i];
+ kernel[len / 2] += 1.0;
}
+ gst_audio_fx_base_fir_filter_set_kernel (GST_AUDIO_FX_BASE_FIR_FILTER (self),
+ kernel, self->kernel_length, (len - 1) / 2);
}
/* GstAudioFilter vmethod implementations */
/* get notified of caps and plug in the correct process function */
static gboolean
-audio_wsinclimit_setup (GstAudioFilter * base, GstRingBufferSpec * format)
+gst_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;
- }
+ gst_audio_wsinclimit_build_kernel (self);
- return GST_BASE_TRANSFORM_CLASS (parent_class)->event (base, event);
+ return GST_AUDIO_FILTER_CLASS (parent_class)->setup (base, format);
}
static void
-audio_wsinclimit_set_property (GObject * object, guint prop_id,
+gst_audio_wsinclimit_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstAudioWSincLimit *self = GST_AUDIO_WSINC_LIMIT (object);
case PROP_LENGTH:{
gint val;
- GST_BASE_TRANSFORM_LOCK (self);
+ GST_OBJECT_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;
- }
+ gst_audio_fx_base_fir_filter_push_residue (GST_AUDIO_FX_BASE_FIR_FILTER
+ (self));
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_audio_wsinclimit_build_kernel (self);
}
- GST_BASE_TRANSFORM_UNLOCK (self);
+ GST_OBJECT_UNLOCK (self);
break;
}
case PROP_FREQUENCY:
- GST_BASE_TRANSFORM_LOCK (self);
+ GST_OBJECT_LOCK (self);
self->cutoff = g_value_get_float (value);
- audio_wsinclimit_build_kernel (self);
- GST_BASE_TRANSFORM_UNLOCK (self);
+ gst_audio_wsinclimit_build_kernel (self);
+ GST_OBJECT_UNLOCK (self);
break;
case PROP_MODE:
- GST_BASE_TRANSFORM_LOCK (self);
+ GST_OBJECT_LOCK (self);
self->mode = g_value_get_enum (value);
- audio_wsinclimit_build_kernel (self);
- GST_BASE_TRANSFORM_UNLOCK (self);
+ gst_audio_wsinclimit_build_kernel (self);
+ GST_OBJECT_UNLOCK (self);
break;
case PROP_WINDOW:
- GST_BASE_TRANSFORM_LOCK (self);
+ GST_OBJECT_LOCK (self);
self->window = g_value_get_enum (value);
- audio_wsinclimit_build_kernel (self);
- GST_BASE_TRANSFORM_UNLOCK (self);
+ gst_audio_wsinclimit_build_kernel (self);
+ GST_OBJECT_UNLOCK (self);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
}
static void
-audio_wsinclimit_get_property (GObject * object, guint prop_id, GValue * value,
- GParamSpec * pspec)
+gst_audio_wsinclimit_get_property (GObject * object, guint prop_id,
+ GValue * value, GParamSpec * pspec)
{
GstAudioWSincLimit *self = GST_AUDIO_WSINC_LIMIT (object);
* GStreamer
* Copyright (C) 1999-2001 Erik Walthinsen <omega@cse.ogi.edu>
* 2006 Dreamlab Technologies Ltd. <mathis.hofer@dreamlab.net>
+ * 2007-2009 Sebastian Dröge <sebastian.droege@collabora.co.uk>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
#include <gst/gst.h>
#include <gst/audio/gstaudiofilter.h>
+#include "audiofxbasefirfilter.h"
+
G_BEGIN_DECLS
#define GST_TYPE_AUDIO_WSINC_LIMIT \
- (audio_wsinclimit_get_type())
+ (gst_audio_wsinclimit_get_type())
#define GST_AUDIO_WSINC_LIMIT(obj) \
(G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIO_WSINC_LIMIT,GstAudioWSincLimit))
#define GST_AUDIO_WSINC_LIMIT_CLASS(klass) \
typedef struct _GstAudioWSincLimit GstAudioWSincLimit;
typedef struct _GstAudioWSincLimitClass GstAudioWSincLimitClass;
-typedef void (*GstAudioWSincLimitProcessFunc) (GstAudioWSincLimit *, guint8 *, guint8 *, guint);
-
/**
* GstAudioWSincLimit:
*
* Opaque data structure.
*/
struct _GstAudioWSincLimit {
- GstAudioFilter element;
+ GstAudioFXBaseFIRFilter parent;
/* < private > */
- GstAudioWSincLimitProcessFunc process;
-
gint mode;
gint window;
gfloat cutoff;
- gint kernel_length; /* length of the filter kernel */
-
- gdouble *residue; /* buffer for left-over samples from previous buffer */
- gdouble *kernel; /* filter kernel */
- gboolean have_kernel;
- gint residue_length;
- guint64 latency;
- GstClockTime next_ts;
- guint64 next_off;
+ gint kernel_length;
};
struct _GstAudioWSincLimitClass {
- GstAudioFilterClass parent_class;
+ GstAudioFXBaseFIRFilterClass parent;
};
-GType audio_wsinclimit_get_type (void);
+GType gst_audio_wsinclimit_get_type (void);
G_END_DECLS
g_object_set (G_OBJECT (audiowsincband), "upper-frequency",
44100 / 4.0 + 1000, NULL);
inbuffer = gst_buffer_new_and_alloc (1024 * sizeof (gfloat));
+ GST_BUFFER_TIMESTAMP (inbuffer) = 0;
in = (gfloat *) GST_BUFFER_DATA (inbuffer);
for (i = 0; i < 1024; i++)
in[i] = 1.0;
g_object_set (G_OBJECT (audiowsincband), "upper-frequency",
44100 / 4.0 + 1000, NULL);
inbuffer = gst_buffer_new_and_alloc (1024 * sizeof (gfloat));
+ GST_BUFFER_TIMESTAMP (inbuffer) = 0;
in = (gfloat *) GST_BUFFER_DATA (inbuffer);
for (i = 0; i < 1024; i += 4) {
in[i] = 0.0;
g_object_set (G_OBJECT (audiowsincband), "upper-frequency",
44100 / 4.0 + 1000, NULL);
inbuffer = gst_buffer_new_and_alloc (1024 * sizeof (gfloat));
+ GST_BUFFER_TIMESTAMP (inbuffer) = 0;
in = (gfloat *) GST_BUFFER_DATA (inbuffer);
for (i = 0; i < 1024; i += 2) {
in[i] = 1.0;
g_object_set (G_OBJECT (audiowsincband), "upper-frequency",
44100 / 4.0 + 1000, NULL);
inbuffer = gst_buffer_new_and_alloc (1024 * sizeof (gfloat));
+ GST_BUFFER_TIMESTAMP (inbuffer) = 0;
in = (gfloat *) GST_BUFFER_DATA (inbuffer);
for (i = 0; i < 1024; i++)
in[i] = 1.0;
g_object_set (G_OBJECT (audiowsincband), "upper-frequency",
44100 / 4.0 + 1000, NULL);
inbuffer = gst_buffer_new_and_alloc (1024 * sizeof (gfloat));
+ GST_BUFFER_TIMESTAMP (inbuffer) = 0;
in = (gfloat *) GST_BUFFER_DATA (inbuffer);
for (i = 0; i < 1024; i += 4) {
g_object_set (G_OBJECT (audiowsincband), "upper-frequency",
44100 / 4.0 + 1000, NULL);
inbuffer = gst_buffer_new_and_alloc (1024 * sizeof (gfloat));
+ GST_BUFFER_TIMESTAMP (inbuffer) = 0;
in = (gfloat *) GST_BUFFER_DATA (inbuffer);
for (i = 0; i < 1024; i += 2) {
in[i] = 1.0;
g_object_set (G_OBJECT (audiowsincband), "upper-frequency",
44100 / 4.0 + 44100 / 16.0, NULL);
inbuffer = gst_buffer_new_and_alloc (20 * sizeof (gfloat));
+ GST_BUFFER_TIMESTAMP (inbuffer) = 0;
in = (gfloat *) GST_BUFFER_DATA (inbuffer);
for (i = 0; i < 20; i++)
in[i] = 1.0;
g_object_set (G_OBJECT (audiowsincband), "upper-frequency",
44100 / 4.0 + 1000, NULL);
inbuffer = gst_buffer_new_and_alloc (1024 * sizeof (gdouble));
+ GST_BUFFER_TIMESTAMP (inbuffer) = 0;
in = (gdouble *) GST_BUFFER_DATA (inbuffer);
for (i = 0; i < 1024; i++)
in[i] = 1.0;
g_object_set (G_OBJECT (audiowsincband), "upper-frequency",
44100 / 4.0 + 1000, NULL);
inbuffer = gst_buffer_new_and_alloc (1024 * sizeof (gdouble));
+ GST_BUFFER_TIMESTAMP (inbuffer) = 0;
in = (gdouble *) GST_BUFFER_DATA (inbuffer);
for (i = 0; i < 1024; i += 4) {
in[i] = 0.0;
g_object_set (G_OBJECT (audiowsincband), "upper-frequency",
44100 / 4.0 + 1000, NULL);
inbuffer = gst_buffer_new_and_alloc (1024 * sizeof (gdouble));
+ GST_BUFFER_TIMESTAMP (inbuffer) = 0;
in = (gdouble *) GST_BUFFER_DATA (inbuffer);
for (i = 0; i < 1024; i += 2) {
in[i] = 1.0;
g_object_set (G_OBJECT (audiowsincband), "upper-frequency",
44100 / 4.0 + 1000, NULL);
inbuffer = gst_buffer_new_and_alloc (1024 * sizeof (gdouble));
+ GST_BUFFER_TIMESTAMP (inbuffer) = 0;
in = (gdouble *) GST_BUFFER_DATA (inbuffer);
for (i = 0; i < 1024; i++)
in[i] = 1.0;
g_object_set (G_OBJECT (audiowsincband), "upper-frequency",
44100 / 4.0 + 1000, NULL);
inbuffer = gst_buffer_new_and_alloc (1024 * sizeof (gdouble));
+ GST_BUFFER_TIMESTAMP (inbuffer) = 0;
in = (gdouble *) GST_BUFFER_DATA (inbuffer);
for (i = 0; i < 1024; i += 4) {
g_object_set (G_OBJECT (audiowsincband), "upper-frequency",
44100 / 4.0 + 1000, NULL);
inbuffer = gst_buffer_new_and_alloc (1024 * sizeof (gdouble));
+ GST_BUFFER_TIMESTAMP (inbuffer) = 0;
in = (gdouble *) GST_BUFFER_DATA (inbuffer);
for (i = 0; i < 1024; i += 2) {
in[i] = 1.0;
g_object_set (G_OBJECT (audiowsincband), "upper-frequency",
44100 / 4.0 + 44100 / 16.0, NULL);
inbuffer = gst_buffer_new_and_alloc (20 * sizeof (gdouble));
+ GST_BUFFER_TIMESTAMP (inbuffer) = 0;
in = (gdouble *) GST_BUFFER_DATA (inbuffer);
for (i = 0; i < 20; i++)
in[i] = 1.0;
/* cutoff = sampling rate / 4, data = 0 */
g_object_set (G_OBJECT (audiowsinclimit), "cutoff", 44100 / 4.0, NULL);
inbuffer = gst_buffer_new_and_alloc (128 * sizeof (gfloat));
+ GST_BUFFER_TIMESTAMP (inbuffer) = 0;
in = (gfloat *) GST_BUFFER_DATA (inbuffer);
for (i = 0; i < 128; i++)
in[i] = 1.0;
g_object_set (G_OBJECT (audiowsinclimit), "cutoff", 44100 / 4.0, NULL);
inbuffer = gst_buffer_new_and_alloc (128 * sizeof (gfloat));
+ GST_BUFFER_TIMESTAMP (inbuffer) = 0;
in = (gfloat *) GST_BUFFER_DATA (inbuffer);
for (i = 0; i < 128; i += 2) {
in[i] = 1.0;
g_object_set (G_OBJECT (audiowsinclimit), "cutoff", 44100 / 4.0, NULL);
inbuffer = gst_buffer_new_and_alloc (128 * sizeof (gfloat));
+ GST_BUFFER_TIMESTAMP (inbuffer) = 0;
in = (gfloat *) GST_BUFFER_DATA (inbuffer);
for (i = 0; i < 128; i++)
in[i] = 1.0;
g_object_set (G_OBJECT (audiowsinclimit), "cutoff", 44100 / 4.0, NULL);
inbuffer = gst_buffer_new_and_alloc (128 * sizeof (gfloat));
+ GST_BUFFER_TIMESTAMP (inbuffer) = 0;
in = (gfloat *) GST_BUFFER_DATA (inbuffer);
for (i = 0; i < 128; i += 2) {
in[i] = 1.0;
g_object_set (G_OBJECT (audiowsinclimit), "cutoff", 44100 / 4.0, NULL);
inbuffer = gst_buffer_new_and_alloc (20 * sizeof (gfloat));
+ GST_BUFFER_TIMESTAMP (inbuffer) = 0;
in = (gfloat *) GST_BUFFER_DATA (inbuffer);
for (i = 0; i < 20; i++)
in[i] = 1.0;
/* cutoff = sampling rate / 4, data = 0 */
g_object_set (G_OBJECT (audiowsinclimit), "cutoff", 44100 / 4.0, NULL);
inbuffer = gst_buffer_new_and_alloc (128 * sizeof (gdouble));
+ GST_BUFFER_TIMESTAMP (inbuffer) = 0;
in = (gdouble *) GST_BUFFER_DATA (inbuffer);
for (i = 0; i < 128; i++)
in[i] = 1.0;
g_object_set (G_OBJECT (audiowsinclimit), "cutoff", 44100 / 4.0, NULL);
inbuffer = gst_buffer_new_and_alloc (128 * sizeof (gdouble));
+ GST_BUFFER_TIMESTAMP (inbuffer) = 0;
in = (gdouble *) GST_BUFFER_DATA (inbuffer);
for (i = 0; i < 128; i += 2) {
in[i] = 1.0;
g_object_set (G_OBJECT (audiowsinclimit), "cutoff", 44100 / 4.0, NULL);
inbuffer = gst_buffer_new_and_alloc (128 * sizeof (gdouble));
+ GST_BUFFER_TIMESTAMP (inbuffer) = 0;
in = (gdouble *) GST_BUFFER_DATA (inbuffer);
for (i = 0; i < 128; i++)
in[i] = 1.0;
g_object_set (G_OBJECT (audiowsinclimit), "cutoff", 44100 / 4.0, NULL);
inbuffer = gst_buffer_new_and_alloc (128 * sizeof (gdouble));
+ GST_BUFFER_TIMESTAMP (inbuffer) = 0;
in = (gdouble *) GST_BUFFER_DATA (inbuffer);
for (i = 0; i < 128; i += 2) {
in[i] = 1.0;
g_object_set (G_OBJECT (audiowsinclimit), "cutoff", 44100 / 4.0, NULL);
inbuffer = gst_buffer_new_and_alloc (20 * sizeof (gdouble));
+ GST_BUFFER_TIMESTAMP (inbuffer) = 0;
in = (gdouble *) GST_BUFFER_DATA (inbuffer);
for (i = 0; i < 20; i++)
in[i] = 1.0;
projects / platform / upstream / gstreamer.git / commitdiff