/* -*- c-basic-offset: 2 -*- * * GStreamer * Copyright (C) 1999-2001 Erik Walthinsen * 2006 Dreamlab Technologies Ltd. * 2007 Sebastian Dröge * * 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. * * * this windowed sinc filter is taken from the freely downloadable DSP book, * "The Scientist and Engineer's Guide to Digital Signal Processing", * chapter 16 * available at http://www.dspguide.com/ * * 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 */ /** * SECTION:element-audiowsinclimit * @short_description: Windowed Sinc low pass and high pass filter * * * * Attenuates all frequencies above the cutoff frequency (low-pass) or all frequencies below the * cutoff frequency (high-pass). The length parameter controls the rolloff, the window parameter * controls rolloff and stopband attenuation. The Hamming window provides a faster rolloff but a bit * worse stopband attenuation, the other way around for the Blackman window. * * * This element has the advantage over the Chebyshev lowpass and highpass filter that it has * a much better rolloff when using a larger kernel size and almost linear phase. The only * disadvantage is the much slower execution time with larger kernels. * * Example launch line * * * gst-launch audiotestsrc freq=1500 ! audioconvert ! audiowsinclimit mode=low-pass frequency=1000 length=501 ! audioconvert ! alsasink * gst-launch filesrc location="melo1.ogg" ! oggdemux ! vorbisdec ! audioconvert ! audiowsinclimit mode=high-pass frequency=15000 length=501 ! audioconvert ! alsasink * gst-launch audiotestsrc wave=white-noise ! audioconvert ! audiowsinclimit mode=low-pass frequency=1000 length=10001 window=blackman ! audioconvert ! alsasink * * * */ #ifdef HAVE_CONFIG_H #include "config.h" #endif #include #include #include #include #include #include "audiowsinclimit.h" #define GST_CAT_DEFAULT audio_wsinclimit_debug GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT); static const GstElementDetails audio_wsinclimit_details = GST_ELEMENT_DETAILS ("AudioWSincLimit", "Filter/Effect/Audio", "Low-pass and High-pass Windowed sinc filter", "Thomas , " "Steven W. Smith, " "Dreamlab Technologies Ltd. , " "Sebastian Dröge "); /* Filter signals and args */ enum { /* FILL ME */ LAST_SIGNAL }; enum { PROP_0, PROP_LENGTH, PROP_FREQUENCY, PROP_MODE, PROP_WINDOW }; enum { MODE_LOW_PASS = 0, MODE_HIGH_PASS }; #define GST_TYPE_AUDIO_WSINC_LIMIT_MODE (audio_wsinclimit_mode_get_type ()) static GType audio_wsinclimit_mode_get_type (void) { static GType gtype = 0; if (gtype == 0) { static const GEnumValue values[] = { {MODE_LOW_PASS, "Low pass (default)", "low-pass"}, {MODE_HIGH_PASS, "High pass", "high-pass"}, {0, NULL, NULL} }; gtype = g_enum_register_static ("GstAudioWSincLimitMode", values); } return gtype; } enum { WINDOW_HAMMING = 0, WINDOW_BLACKMAN }; #define GST_TYPE_AUDIO_WSINC_LIMIT_WINDOW (audio_wsinclimit_window_get_type ()) static GType audio_wsinclimit_window_get_type (void) { static GType gtype = 0; if (gtype == 0) { static const GEnumValue values[] = { {WINDOW_HAMMING, "Hamming window (default)", "hamming"}, {WINDOW_BLACKMAN, "Blackman window", "blackman"}, {0, NULL, NULL} }; gtype = g_enum_register_static ("GstAudioWSincLimitWindow", values); } 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, \ "Low-pass and High-pass Windowed sinc filter plugin"); GST_BOILERPLATE_FULL (GstAudioWSincLimit, audio_wsinclimit, GstAudioFilter, GST_TYPE_AUDIO_FILTER, DEBUG_INIT); static void 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, 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, 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) { 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); } static void 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; /* 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_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)); 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)); 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)); 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); } static void 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) { gint i = 0; gdouble sum = 0.0; gint len = 0; gdouble w; len = self->kernel_length; if (GST_AUDIO_FILTER (self)->format.rate == 0) { GST_DEBUG ("rate not set yet"); return; } if (GST_AUDIO_FILTER (self)->format.channels == 0) { GST_DEBUG ("channels not set yet"); return; } /* Clamp cutoff frequency between 0 and the nyquist frequency */ self->cutoff = CLAMP (self->cutoff, 0.0, GST_AUDIO_FILTER (self)->format.rate / 2); GST_DEBUG ("audio_wsinclimit_: initializing filter kernel of length %d " "with cutoff %.2lf Hz " "for mode %s", len, self->cutoff, (self->mode == MODE_LOW_PASS) ? "low-pass" : "high-pass"); /* 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); for (i = 0; i < len; ++i) { if (i == len / 2) self->kernel[i] = w; else self->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)); else self->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]; for (i = 0; i < len; ++i) self->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"); } } /* GstAudioFilter vmethod implementations */ /* get notified of caps and plug in the correct process function */ static gboolean audio_wsinclimit_setup (GstAudioFilter * base, GstRingBufferSpec * format) { GstAudioWSincLimit *self = GST_AUDIO_WSINC_LIMIT (base); gboolean ret = TRUE; if (format->width == 32) self->process = (GstAudioWSincLimitProcessFunc) process_32; else if (format->width == 64) self->process = (GstAudioWSincLimitProcessFunc) process_64; else ret = FALSE; self->have_kernel = FALSE; return TRUE; } /* GstBaseTransform vmethod implementations */ static GstFlowReturn audio_wsinclimit_transform (GstBaseTransform * base, GstBuffer * inbuf, GstBuffer * outbuf) { GstAudioWSincLimit *self = GST_AUDIO_WSINC_LIMIT (base); GstClockTime timestamp; gint channels = GST_AUDIO_FILTER (self)->format.channels; gint rate = GST_AUDIO_FILTER (self)->format.rate; gint input_samples = GST_BUFFER_SIZE (outbuf) / (GST_AUDIO_FILTER (self)->format.width / 8); gint output_samples = input_samples; gint diff; /* FIXME: subdivide GST_BUFFER_SIZE into small chunks for smooth fades */ timestamp = GST_BUFFER_TIMESTAMP (outbuf); if (GST_CLOCK_TIME_IS_VALID (timestamp)) gst_object_sync_values (G_OBJECT (self), timestamp); if (!self->have_kernel) audio_wsinclimit_build_kernel (self); /* Reset the residue if already existing on discont buffers */ if (GST_BUFFER_IS_DISCONT (inbuf)) { if (channels && self->residue) memset (self->residue, 0, channels * self->kernel_length * sizeof (gdouble)); self->residue_length = 0; self->next_ts = GST_CLOCK_TIME_NONE; self->next_off = GST_BUFFER_OFFSET_NONE; } /* Calculate the number of samples we can push out now without outputting * kernel_length/2 zeros in the beginning */ diff = (self->kernel_length / 2) * channels - self->residue_length; if (diff > 0) output_samples -= diff; self->process (self, GST_BUFFER_DATA (inbuf), GST_BUFFER_DATA (outbuf), input_samples); if (output_samples <= 0) { /* Drop buffer and save original timestamp/offset for later use */ if (!GST_CLOCK_TIME_IS_VALID (self->next_ts) && GST_BUFFER_TIMESTAMP_IS_VALID (outbuf)) self->next_ts = GST_BUFFER_TIMESTAMP (outbuf); if (self->next_off == GST_BUFFER_OFFSET_NONE && GST_BUFFER_OFFSET_IS_VALID (outbuf)) self->next_off = GST_BUFFER_OFFSET (outbuf); return GST_BASE_TRANSFORM_FLOW_DROPPED; } else if (output_samples < input_samples) { /* First (probably partial) buffer after starting from * a clean residue. Use stored timestamp/offset here */ if (GST_CLOCK_TIME_IS_VALID (self->next_ts)) GST_BUFFER_TIMESTAMP (outbuf) = self->next_ts; if (self->next_off != GST_BUFFER_OFFSET_NONE) { GST_BUFFER_OFFSET (outbuf) = self->next_off; if (GST_BUFFER_OFFSET_END_IS_VALID (outbuf)) GST_BUFFER_OFFSET_END (outbuf) = self->next_off + output_samples / channels; } else { /* We dropped no buffer, offset is valid, offset_end must be adjusted by diff */ if (GST_BUFFER_OFFSET_END_IS_VALID (outbuf)) GST_BUFFER_OFFSET_END (outbuf) -= diff / channels; } if (GST_BUFFER_DURATION_IS_VALID (outbuf)) GST_BUFFER_DURATION (outbuf) -= gst_util_uint64_scale (diff, GST_SECOND, channels * rate); GST_BUFFER_DATA (outbuf) += diff * (GST_AUDIO_FILTER (self)->format.width / 8); GST_BUFFER_SIZE (outbuf) -= diff * (GST_AUDIO_FILTER (self)->format.width / 8); } else { GstClockTime ts_latency = gst_util_uint64_scale (self->latency, GST_SECOND, rate); /* Normal buffer, adjust timestamp/offset/etc by latency */ if (GST_BUFFER_TIMESTAMP (outbuf) < ts_latency) { GST_WARNING_OBJECT (self, "GST_BUFFER_TIMESTAMP (outbuf) < latency"); GST_BUFFER_TIMESTAMP (outbuf) = 0; } else { GST_BUFFER_TIMESTAMP (outbuf) -= ts_latency; } if (GST_BUFFER_OFFSET_IS_VALID (outbuf)) { if (GST_BUFFER_OFFSET (outbuf) > self->latency) { GST_BUFFER_OFFSET (outbuf) -= self->latency; } else { GST_WARNING_OBJECT (self, "GST_BUFFER_OFFSET (outbuf) < latency"); GST_BUFFER_OFFSET (outbuf) = 0; } } if (GST_BUFFER_OFFSET_END_IS_VALID (outbuf)) { if (GST_BUFFER_OFFSET_END (outbuf) > self->latency) { GST_BUFFER_OFFSET_END (outbuf) -= self->latency; } else { GST_WARNING_OBJECT (self, "GST_BUFFER_OFFSET_END (outbuf) < latency"); GST_BUFFER_OFFSET_END (outbuf) = 0; } } } GST_DEBUG_OBJECT (self, "Pushing buffer of size %d with timestamp: %" GST_TIME_FORMAT ", duration: %" GST_TIME_FORMAT ", offset: %lld," " offset_end: %lld, nsamples: %d", GST_BUFFER_SIZE (outbuf), GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf)), GST_TIME_ARGS (GST_BUFFER_DURATION (outbuf)), GST_BUFFER_OFFSET (outbuf), GST_BUFFER_OFFSET_END (outbuf), output_samples / channels); self->next_ts = GST_BUFFER_TIMESTAMP (outbuf) + GST_BUFFER_DURATION (outbuf); self->next_off = GST_BUFFER_OFFSET_END (outbuf); return GST_FLOW_OK; } static gboolean audio_wsinclimit_start (GstBaseTransform * base) { GstAudioWSincLimit *self = GST_AUDIO_WSINC_LIMIT (base); gint channels = GST_AUDIO_FILTER (self)->format.channels; /* Reset the residue if already existing */ if (channels && self->residue) memset (self->residue, 0, channels * self->kernel_length * sizeof (gdouble)); self->residue_length = 0; self->next_ts = GST_CLOCK_TIME_NONE; self->next_off = GST_BUFFER_OFFSET_NONE; return TRUE; } static gboolean audio_wsinclimit_query (GstPad * pad, GstQuery * query) { GstAudioWSincLimit *self = GST_AUDIO_WSINC_LIMIT (gst_pad_get_parent (pad)); gboolean res = TRUE; switch (GST_QUERY_TYPE (query)) { case GST_QUERY_LATENCY: { GstClockTime min, max; gboolean live; guint64 latency; GstPad *peer; gint rate = GST_AUDIO_FILTER (self)->format.rate; if ((peer = gst_pad_get_peer (GST_BASE_TRANSFORM (self)->sinkpad))) { if ((res = gst_pad_query (peer, query))) { gst_query_parse_latency (query, &live, &min, &max); GST_DEBUG_OBJECT (self, "Peer latency: min %" GST_TIME_FORMAT " max %" GST_TIME_FORMAT, GST_TIME_ARGS (min), GST_TIME_ARGS (max)); /* add our own latency */ latency = (rate != 0) ? gst_util_uint64_scale (self->latency, GST_SECOND, rate) : 0; GST_DEBUG_OBJECT (self, "Our latency: %" GST_TIME_FORMAT, GST_TIME_ARGS (latency)); min += latency; if (max != GST_CLOCK_TIME_NONE) max += latency; GST_DEBUG_OBJECT (self, "Calculated total latency : min %" GST_TIME_FORMAT " max %" GST_TIME_FORMAT, GST_TIME_ARGS (min), GST_TIME_ARGS (max)); gst_query_set_latency (query, live, min, max); } gst_object_unref (peer); } break; } default: res = gst_pad_query_default (pad, query); break; } gst_object_unref (self); return res; } static const GstQueryType * audio_wsinclimit_query_type (GstPad * pad) { static const GstQueryType types[] = { GST_QUERY_LATENCY, 0 }; return types; } static gboolean audio_wsinclimit_event (GstBaseTransform * base, GstEvent * event) { GstAudioWSincLimit *self = GST_AUDIO_WSINC_LIMIT (base); switch (GST_EVENT_TYPE (event)) { case GST_EVENT_EOS: audio_wsinclimit_push_residue (self); break; default: break; } return GST_BASE_TRANSFORM_CLASS (parent_class)->event (base, event); } static void audio_wsinclimit_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec) { GstAudioWSincLimit *self = GST_AUDIO_WSINC_LIMIT (object); g_return_if_fail (GST_IS_AUDIO_WSINC_LIMIT (self)); switch (prop_id) { case PROP_LENGTH:{ gint val; GST_BASE_TRANSFORM_LOCK (self); val = g_value_get_int (value); if (val % 2 == 0) val++; if (val != self->kernel_length) { if (self->residue) { audio_wsinclimit_push_residue (self); g_free (self->residue); self->residue = NULL; } self->kernel_length = val; self->latency = val / 2; audio_wsinclimit_build_kernel (self); gst_element_post_message (GST_ELEMENT (self), gst_message_new_latency (GST_OBJECT (self))); } GST_BASE_TRANSFORM_UNLOCK (self); break; } case PROP_FREQUENCY: GST_BASE_TRANSFORM_LOCK (self); self->cutoff = g_value_get_float (value); audio_wsinclimit_build_kernel (self); GST_BASE_TRANSFORM_UNLOCK (self); break; case PROP_MODE: GST_BASE_TRANSFORM_LOCK (self); self->mode = g_value_get_enum (value); audio_wsinclimit_build_kernel (self); GST_BASE_TRANSFORM_UNLOCK (self); break; case PROP_WINDOW: GST_BASE_TRANSFORM_LOCK (self); self->window = g_value_get_enum (value); audio_wsinclimit_build_kernel (self); GST_BASE_TRANSFORM_UNLOCK (self); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } } static void audio_wsinclimit_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec) { GstAudioWSincLimit *self = GST_AUDIO_WSINC_LIMIT (object); switch (prop_id) { case PROP_LENGTH: g_value_set_int (value, self->kernel_length); break; case PROP_FREQUENCY: g_value_set_float (value, self->cutoff); break; case PROP_MODE: g_value_set_enum (value, self->mode); break; case PROP_WINDOW: g_value_set_enum (value, self->window); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } }