diff options
Diffstat (limited to 'gst')
| -rw-r--r-- | gst/filter/Makefile.am | 4 | ||||
| -rw-r--r-- | gst/filter/filter.vcproj | 6 | ||||
| -rw-r--r-- | gst/filter/gstbpwsinc.c | 863 | ||||
| -rw-r--r-- | gst/filter/gstbpwsinc.h | 88 | ||||
| -rw-r--r-- | gst/filter/gstfilter.c | 4 | ||||
| -rw-r--r-- | gst/filter/gstfilter.h | 2 | ||||
| -rw-r--r-- | gst/filter/gstlpwsinc.c | 795 | ||||
| -rw-r--r-- | gst/filter/gstlpwsinc.h | 88 |
8 files changed, 3 insertions, 1847 deletions
diff --git a/gst/filter/Makefile.am b/gst/filter/Makefile.am index e4f27784..35132517 100644 --- a/gst/filter/Makefile.am +++ b/gst/filter/Makefile.am @@ -1,7 +1,7 @@ plugin_LTLIBRARIES = libgstfilter.la -libgstfilter_la_SOURCES = gstfilter.c gstlpwsinc.c gstbpwsinc.c gstiir.c iir.c +libgstfilter_la_SOURCES = gstfilter.c gstiir.c iir.c libgstfilter_la_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(GST_BASE_CFLAGS) $(GST_CFLAGS) $(GST_CONTROLLER_CFLAGS) libgstfilter_la_LIBADD = \ $(GST_BASE_LIBS) \ @@ -13,4 +13,4 @@ libgstfilter_la_LIBADD = \ libgstfilter_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS) -noinst_HEADERS = gstfilter.h gstlpwsinc.h gstbpwsinc.h gstiir.h iir.h +noinst_HEADERS = gstfilter.h gstiir.h iir.h diff --git a/gst/filter/filter.vcproj b/gst/filter/filter.vcproj index 59434e85..5b8f3124 100644 --- a/gst/filter/filter.vcproj +++ b/gst/filter/filter.vcproj @@ -134,12 +134,6 @@ <File RelativePath=".\iir.c"> </File> - <File - RelativePath=".\gstlpwsinc.c"> - </File> - <File - RelativePath=".\gstbpwsinc.c"> - </File> </Filter> <Filter Name="Header Files" diff --git a/gst/filter/gstbpwsinc.c b/gst/filter/gstbpwsinc.c deleted file mode 100644 index 929daec6..00000000 --- a/gst/filter/gstbpwsinc.c +++ /dev/null @@ -1,863 +0,0 @@ -/* -*- c-basic-offset: 2 -*- - * - * 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> - * - * 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 - * - Drop the first kernel_length/2 samples and append the same number of - * samples on EOS as the first few samples are essentialy zero. - */ - -/** - * SECTION:element-bpwsinc - * @short_description: Windowed Sinc band pass and band reject filter - * - * <refsect2> - * <para> - * Attenuates all frequencies outside (bandpass) or inside (bandreject) of a frequency - * band. 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. - * </para> - * <para> - * This element has the advantage over the Chebyshev bandpass and bandreject 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. - * </para> - * <title>Example launch line</title> - * <para> - * <programlisting> - * gst-launch audiotestsrc freq=1500 ! audioconvert ! bpwsinc mode=band-pass lower-frequency=3000 upper-frequency=10000 length=501 window=blackman ! audioconvert ! alsasink - * gst-launch filesrc location="melo1.ogg" ! oggdemux ! vorbisdec ! audioconvert ! bpwsinc mode=band-reject lower-frequency=59 upper-frequency=61 length=10001 window=hamming ! audioconvert ! alsasink - * gst-launch audiotestsrc wave=white-noise ! audioconvert ! bpwsinc mode=band-pass lower-frequency=1000 upper-frequency=2000 length=31 ! audioconvert ! alsasink - * </programlisting> - * </para> - * </refsect2> - */ - -#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 "gstbpwsinc.h" - -#define GST_CAT_DEFAULT gst_bpwsinc_debug -GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT); - -static const GstElementDetails bpwsinc_details = -GST_ELEMENT_DETAILS ("Band-pass and Band-reject Windowed sinc filter", - "Filter/Effect/Audio", - "Band-pass Windowed sinc filter", - "Thomas <thomas@apestaart.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, - PROP_LENGTH, - PROP_LOWER_FREQUENCY, - PROP_UPPER_FREQUENCY, - PROP_MODE, - PROP_WINDOW -}; - -enum -{ - MODE_BAND_PASS = 0, - MODE_BAND_REJECT -}; - -#define GST_TYPE_BPWSINC_MODE (gst_bpwsinc_mode_get_type ()) -static GType -gst_bpwsinc_mode_get_type (void) -{ - static GType gtype = 0; - - if (gtype == 0) { - static const GEnumValue values[] = { - {MODE_BAND_PASS, "Band pass (default)", - "band-pass"}, - {MODE_BAND_REJECT, "Band reject", - "band-reject"}, - {0, NULL, NULL} - }; - - gtype = g_enum_register_static ("GstBPWSincMode", values); - } - return gtype; -} - -enum -{ - WINDOW_HAMMING = 0, - WINDOW_BLACKMAN -}; - -#define GST_TYPE_BPWSINC_WINDOW (gst_bpwsinc_window_get_type ()) -static GType -gst_bpwsinc_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 ("GstBPWSincWindow", 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 (gst_bpwsinc_debug, "bpwsinc", 0, "Band-pass and Band-reject Windowed sinc filter plugin"); - -GST_BOILERPLATE_FULL (GstBPWSinc, gst_bpwsinc, GstAudioFilter, - GST_TYPE_AUDIO_FILTER, DEBUG_INIT); - -static void bpwsinc_set_property (GObject * object, guint prop_id, - const GValue * value, GParamSpec * pspec); -static void bpwsinc_get_property (GObject * object, guint prop_id, - GValue * value, GParamSpec * pspec); - -static GstFlowReturn bpwsinc_transform (GstBaseTransform * base, - GstBuffer * inbuf, GstBuffer * outbuf); -static gboolean bpwsinc_start (GstBaseTransform * base); -static gboolean bpwsinc_event (GstBaseTransform * base, GstEvent * event); - -static gboolean bpwsinc_setup (GstAudioFilter * base, - GstRingBufferSpec * format); - -static gboolean bpwsinc_query (GstPad * pad, GstQuery * query); -static const GstQueryType *bpwsinc_query_type (GstPad * pad); - -/* Element class */ - -static void -gst_bpwsinc_dispose (GObject * object) -{ - GstBPWSinc *self = GST_BPWSINC (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_bpwsinc_base_init (gpointer g_class) -{ - GstElementClass *element_class = GST_ELEMENT_CLASS (g_class); - GstCaps *caps; - - gst_element_class_set_details (element_class, &bpwsinc_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 -gst_bpwsinc_class_init (GstBPWSincClass * 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 = bpwsinc_set_property; - gobject_class->get_property = bpwsinc_get_property; - gobject_class->dispose = gst_bpwsinc_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_LOWER_FREQUENCY, - g_param_spec_float ("lower-frequency", "Lower Frequency", - "Cut-off lower frequency (Hz)", 0.0, 100000.0, 0, G_PARAM_READWRITE)); - 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)); - 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)); - - g_object_class_install_property (gobject_class, PROP_MODE, - g_param_spec_enum ("mode", "Mode", - "Band pass or band reject mode", GST_TYPE_BPWSINC_MODE, - MODE_BAND_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_BPWSINC_WINDOW, - WINDOW_HAMMING, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE)); - - trans_class->transform = GST_DEBUG_FUNCPTR (bpwsinc_transform); - trans_class->start = GST_DEBUG_FUNCPTR (bpwsinc_start); - trans_class->event = GST_DEBUG_FUNCPTR (bpwsinc_event); - filter_class->setup = GST_DEBUG_FUNCPTR (bpwsinc_setup); -} - -static void -gst_bpwsinc_init (GstBPWSinc * self, GstBPWSincClass * 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, bpwsinc_query); - gst_pad_set_query_type_function (GST_BASE_TRANSFORM (self)->srcpad, - bpwsinc_query_type); -} - -#define DEFINE_PROCESS_FUNC(width,ctype) \ -static void \ -process_##width (GstBPWSinc * 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 -bpwsinc_build_kernel (GstBPWSinc * self) -{ - gint i = 0; - gdouble sum = 0.0; - gint len = 0; - gdouble *kernel_lp, *kernel_hp; - 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 frequencies */ - self->lower_frequency = - CLAMP (self->lower_frequency, 0.0, - GST_AUDIO_FILTER (self)->format.rate / 2); - self->upper_frequency = - CLAMP (self->upper_frequency, 0.0, - GST_AUDIO_FILTER (self)->format.rate / 2); - if (self->lower_frequency > self->upper_frequency) { - gint tmp = self->lower_frequency; - - self->lower_frequency = self->upper_frequency; - self->upper_frequency = tmp; - } - - GST_DEBUG ("bpwsinc: 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, - (self->mode == MODE_BAND_PASS) ? "band-pass" : "band-reject"); - - /* fill the lp kernel */ - w = 2 * M_PI * (self->lower_frequency / GST_AUDIO_FILTER (self)->format.rate); - kernel_lp = g_new (gdouble, len); - for (i = 0; i < len; ++i) { - if (i == len / 2) - kernel_lp[i] = w; - else - kernel_lp[i] = sin (w * (i - len / 2)) - / (i - len / 2); - /* Windowing */ - if (self->window == WINDOW_HAMMING) - kernel_lp[i] *= (0.54 - 0.46 * cos (2 * M_PI * i / len)); - else - kernel_lp[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 */ - sum = 0.0; - for (i = 0; i < len; ++i) - sum += kernel_lp[i]; - for (i = 0; i < len; ++i) - kernel_lp[i] /= sum; - - /* fill the hp kernel */ - w = 2 * M_PI * (self->upper_frequency / GST_AUDIO_FILTER (self)->format.rate); - kernel_hp = g_new (gdouble, len); - for (i = 0; i < len; ++i) { - if (i == len / 2) - kernel_hp[i] = w; - else - kernel_hp[i] = sin (w * (i - len / 2)) - / (i - len / 2); - /* Windowing */ - if (self->window == WINDOW_HAMMING) - kernel_hp[i] *= (0.54 - 0.46 * cos (2 * M_PI * i / len)); - else - kernel_hp[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 */ - sum = 0.0; - for (i = 0; i < len; ++i) - sum += kernel_hp[i]; - for (i = 0; i < len; ++i) - kernel_hp[i] /= sum; - - /* do spectral inversion to go from lowpass to highpass */ - for (i = 0; i < len; ++i) - kernel_hp[i] = -kernel_hp[i]; - kernel_hp[len / 2] += 1; - - /* combine the two kernels */ - if (self->kernel) - g_free (self->kernel); - self->kernel = g_new (gdouble, len); - - for (i = 0; i < len; ++i) - self->kernel[i] = kernel_lp[i] + kernel_hp[i]; - - /* free the helper kernels */ - g_free (kernel_lp); - g_free (kernel_hp); - - /* do spectral inversion to go from bandreject to bandpass - * 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 -bpwsinc_push_residue (GstBPWSinc * 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 -bpwsinc_setup (GstAudioFilter * base, GstRingBufferSpec * format) -{ - GstBPWSinc *self = GST_BPWSINC (base); - - gboolean ret = TRUE; - - if (format->width == 32) - self->process = (GstBPWSincProcessFunc) process_32; - else if (format->width == 64) - self->process = (GstBPWSincProcessFunc) process_64; - else - ret = FALSE; - - self->have_kernel = FALSE; - - return TRUE; -} - -/* GstBaseTransform vmethod implementations */ - -static GstFlowReturn -bpwsinc_transform (GstBaseTransform * base, GstBuffer * inbuf, - GstBuffer * outbuf) -{ - GstBPWSinc *self = GST_BPWSINC (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) - bpwsinc_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 -bpwsinc_start (GstBaseTransform * base) -{ - GstBPWSinc *self = GST_BPWSINC (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 -bpwsinc_query (GstPad * pad, GstQuery * query) -{ - GstBPWSinc *self = GST_BPWSINC (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 * -bpwsinc_query_type (GstPad * pad) -{ - static const GstQueryType types[] = { - GST_QUERY_LATENCY, - 0 - }; - - return types; -} - -static gboolean -bpwsinc_event (GstBaseTransform * base, GstEvent * event) -{ - GstBPWSinc *self = GST_BPWSINC (base); - - switch (GST_EVENT_TYPE (event)) { - case GST_EVENT_EOS: - bpwsinc_push_residue (self); - break; - default: - break; - } - - return GST_BASE_TRANSFORM_CLASS (parent_class)->event (base, event); -} - -static void -bpwsinc_set_property (GObject * object, guint prop_id, const GValue * value, - GParamSpec * pspec) -{ - GstBPWSinc *self = GST_BPWSINC (object); - - g_return_if_fail (GST_IS_BPWSINC (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) { - bpwsinc_push_residue (self); - g_free (self->residue); - self->residue = NULL; - } - self->kernel_length = val; - self->latency = val / 2; - bpwsinc_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_LOWER_FREQUENCY: - GST_BASE_TRANSFORM_LOCK (self); - self->lower_frequency = g_value_get_float (value); - bpwsinc_build_kernel (self); - GST_BASE_TRANSFORM_UNLOCK (self); - break; - case PROP_UPPER_FREQUENCY: - GST_BASE_TRANSFORM_LOCK (self); - self->upper_frequency = g_value_get_float (value); - bpwsinc_build_kernel (self); - GST_BASE_TRANSFORM_UNLOCK (self); - break; - case PROP_MODE: - GST_BASE_TRANSFORM_LOCK (self); - self->mode = g_value_get_enum (value); - bpwsinc_build_kernel (self); - GST_BASE_TRANSFORM_UNLOCK (self); - break; - case PROP_WINDOW: - GST_BASE_TRANSFORM_LOCK (self); - self->window = g_value_get_enum (value); - bpwsinc_build_kernel (self); - GST_BASE_TRANSFORM_UNLOCK (self); - break; - default: - G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); - break; - } -} - -static void -bpwsinc_get_property (GObject * object, guint prop_id, GValue * value, - GParamSpec * pspec) -{ - GstBPWSinc *self = GST_BPWSINC (object); - - switch (prop_id) { - case PROP_LENGTH: - g_value_set_int (value, self->kernel_length); - break; - case PROP_LOWER_FREQUENCY: - g_value_set_float (value, self->lower_frequency); - break; - case PROP_UPPER_FREQUENCY: - g_value_set_float (value, self->upper_frequency); - 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; - } -} diff --git a/gst/filter/gstbpwsinc.h b/gst/filter/gstbpwsinc.h deleted file mode 100644 index 24a4d2da..00000000 --- a/gst/filter/gstbpwsinc.h +++ /dev/null @@ -1,88 +0,0 @@ -/* -*- c-basic-offset: 2 -*- - * - * GStreamer - * Copyright (C) 1999-2001 Erik Walthinsen <omega@cse.ogi.edu> - * 2006 Dreamlab Technologies Ltd. <mathis.hofer@dreamlab.net> - * - * 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/ - * - */ - -#ifndef __GST_BPWSINC_H__ -#define __GST_BPWSINC_H__ - -#include "gstfilter.h" -#include <gst/gst.h> -#include <gst/audio/gstaudiofilter.h> - -G_BEGIN_DECLS - -#define GST_TYPE_BPWSINC \ - (gst_bpwsinc_get_type()) -#define GST_BPWSINC(obj) \ - (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_BPWSINC,GstBPWSinc)) -#define GST_BPWSINC_CLASS(klass) \ - (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_BPWSINC,GstBPWSincClass)) -#define GST_IS_BPWSINC(obj) \ - (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_BPWSINC)) -#define GST_IS_BPWSINC_CLASS(klass) \ - (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_BPWSINC)) - -typedef struct _GstBPWSinc GstBPWSinc; -typedef struct _GstBPWSincClass GstBPWSincClass; - -typedef void (*GstBPWSincProcessFunc) (GstBPWSinc *, guint8 *, guint8 *, guint); - -/** - * GstBPWSinc: - * - * Opaque data structure. - */ -struct _GstBPWSinc { - GstAudioFilter element; - - /* < private > */ - GstBPWSincProcessFunc 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 _GstBPWSincClass { - GstAudioFilterClass parent_class; -}; - -GType gst_bpwsinc_get_type (void); - -G_END_DECLS - -#endif /* __GST_BPWSINC_H__ */ diff --git a/gst/filter/gstfilter.c b/gst/filter/gstfilter.c index 36c37ec4..11c43cc5 100644 --- a/gst/filter/gstfilter.c +++ b/gst/filter/gstfilter.c @@ -38,8 +38,6 @@ struct _elements_entry static struct _elements_entry _elements[] = { {"iir", gst_iir_get_type}, - {"lpwsinc", gst_lpwsinc_get_type}, - {"bpwsinc", gst_bpwsinc_get_type}, {NULL, 0}, }; @@ -65,5 +63,5 @@ plugin_init (GstPlugin * plugin) GST_PLUGIN_DEFINE (GST_VERSION_MAJOR, GST_VERSION_MINOR, "filter", - "IIR, lpwsinc and bpwsinc audio filter elements", + "IIR audio filter element", plugin_init, VERSION, "LGPL", GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN); diff --git a/gst/filter/gstfilter.h b/gst/filter/gstfilter.h index d8e43cb0..d676fa3b 100644 --- a/gst/filter/gstfilter.h +++ b/gst/filter/gstfilter.h @@ -29,7 +29,5 @@ #include <gst/gst.h> GType gst_iir_get_type (void); -GType gst_lpwsinc_get_type (void); -GType gst_bpwsinc_get_type (void); #endif /* __GST_FILTER_H__ */ diff --git a/gst/filter/gstlpwsinc.c b/gst/filter/gstlpwsinc.c deleted file mode 100644 index 7189aaa7..00000000 --- a/gst/filter/gstlpwsinc.c +++ /dev/null @@ -1,795 +0,0 @@ -/* -*- c-basic-offset: 2 -*- - * - * 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> - * - * 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-lpwsinc - * @short_description: Windowed Sinc low pass and high pass filter - * - * <refsect2> - * <para> - * 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. - * </para> - * <para> - * 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. - * </para> - * <title>Example launch line</title> - * <para> - * <programlisting> - * gst-launch audiotestsrc freq=1500 ! audioconvert ! lpwsinc mode=low-pass frequency=1000 length=501 ! audioconvert ! alsasink - * gst-launch filesrc location="melo1.ogg" ! oggdemux ! vorbisdec ! audioconvert ! lpwsinc mode=high-pass frequency=15000 length=501 ! audioconvert ! alsasink - * gst-launch audiotestsrc wave=white-noise ! audioconvert ! lpwsinc mode=low-pass frequency=1000 length=10001 window=blackman ! audioconvert ! alsasink - * </programlisting> - * </para> - * </refsect2> - */ - -#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 "gstlpwsinc.h" - -#define GST_CAT_DEFAULT gst_lpwsinc_debug -GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT); - -static const GstElementDetails lpwsinc_details = GST_ELEMENT_DETAILS ("LPWSinc", - "Filter/Effect/Audio", - "Low-pass and High-pass Windowed sinc filter", - "Thomas <thomas@apestaart.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, - PROP_LENGTH, - PROP_FREQUENCY, - PROP_MODE, - PROP_WINDOW -}; - -enum -{ - MODE_LOW_PASS = 0, - MODE_HIGH_PASS -}; - -#define GST_TYPE_LPWSINC_MODE (gst_lpwsinc_mode_get_type ()) -static GType -gst_lpwsinc_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 ("GstLPWSincMode", values); - } - return gtype; -} - -enum -{ - WINDOW_HAMMING = 0, - WINDOW_BLACKMAN -}; - -#define GST_TYPE_LPWSINC_WINDOW (gst_lpwsinc_window_get_type ()) -static GType -gst_lpwsinc_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 ("GstLPWSincWindow", 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 (gst_lpwsinc_debug, "lpwsinc", 0, "Low-pass and High-pass Windowed sinc filter plugin"); - -GST_BOILERPLATE_FULL (GstLPWSinc, gst_lpwsinc, GstAudioFilter, - GST_TYPE_AUDIO_FILTER, DEBUG_INIT); - -static void lpwsinc_set_property (GObject * object, guint prop_id, - const GValue * value, GParamSpec * pspec); -static void lpwsinc_get_property (GObject * object, guint prop_id, - GValue * value, GParamSpec * pspec); - -static GstFlowReturn lpwsinc_transform (GstBaseTransform * base, - GstBuffer * inbuf, GstBuffer * outbuf); -static gboolean lpwsinc_start (GstBaseTransform * base); -static gboolean lpwsinc_event (GstBaseTransform * base, GstEvent * event); -static gboolean lpwsinc_setup (GstAudioFilter * base, - GstRingBufferSpec * format); - -static gboolean lpwsinc_query (GstPad * pad, GstQuery * query); -static const GstQueryType *lpwsinc_query_type (GstPad * pad); - -/* Element class */ - -static void -gst_lpwsinc_dispose (GObject * object) -{ - GstLPWSinc *self = GST_LPWSINC (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_lpwsinc_base_init (gpointer g_class) -{ - GstElementClass *element_class = GST_ELEMENT_CLASS (g_class); - GstCaps *caps; - - gst_element_class_set_details (element_class, &lpwsinc_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 -gst_lpwsinc_class_init (GstLPWSincClass * 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 = lpwsinc_set_property; - gobject_class->get_property = lpwsinc_get_property; - gobject_class->dispose = gst_lpwsinc_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_LPWSINC_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_LPWSINC_WINDOW, - WINDOW_HAMMING, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE)); - - trans_class->transform = GST_DEBUG_FUNCPTR (lpwsinc_transform); - trans_class->start = GST_DEBUG_FUNCPTR (lpwsinc_start); - trans_class->event = GST_DEBUG_FUNCPTR (lpwsinc_event); - filter_class->setup = GST_DEBUG_FUNCPTR (lpwsinc_setup); -} - -static void -gst_lpwsinc_init (GstLPWSinc * self, GstLPWSincClass * 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, lpwsinc_query); - gst_pad_set_query_type_function (GST_BASE_TRANSFORM (self)->srcpad, - lpwsinc_query_type); -} - -#define DEFINE_PROCESS_FUNC(width,ctype) \ -static void \ -process_##width (GstLPWSinc * 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 -lpwsinc_build_kernel (GstLPWSinc * 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 ("lpwsinc: 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 -lpwsinc_push_residue (GstLPWSinc * 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 -lpwsinc_setup (GstAudioFilter * base, GstRingBufferSpec * format) -{ - GstLPWSinc *self = GST_LPWSINC (base); - - gboolean ret = TRUE; - - if (format->width == 32) - self->process = (GstLPWSincProcessFunc) process_32; - else if (format->width == 64) - self->process = (GstLPWSincProcessFunc) process_64; - else - ret = FALSE; - - self->have_kernel = FALSE; - - return TRUE; -} - -/* GstBaseTransform vmethod implementations */ - -static GstFlowReturn -lpwsinc_transform (GstBaseTransform * base, GstBuffer * inbuf, - GstBuffer * outbuf) -{ - GstLPWSinc *self = GST_LPWSINC (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) - lpwsinc_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 -lpwsinc_start (GstBaseTransform * base) -{ - GstLPWSinc *self = GST_LPWSINC (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 -lpwsinc_query (GstPad * pad, GstQuery * query) -{ - GstLPWSinc *self = GST_LPWSINC (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 * -lpwsinc_query_type (GstPad * pad) -{ - static const GstQueryType types[] = { - GST_QUERY_LATENCY, - 0 - }; - - return types; -} - -static gboolean -lpwsinc_event (GstBaseTransform * base, GstEvent * event) -{ - GstLPWSinc *self = GST_LPWSINC (base); - - switch (GST_EVENT_TYPE (event)) { - case GST_EVENT_EOS: - lpwsinc_push_residue (self); - break; - default: - break; - } - - return GST_BASE_TRANSFORM_CLASS (parent_class)->event (base, event); -} - -static void -lpwsinc_set_property (GObject * object, guint prop_id, const GValue * value, - GParamSpec * pspec) -{ - GstLPWSinc *self = GST_LPWSINC (object); - - g_return_if_fail (GST_IS_LPWSINC (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) { - lpwsinc_push_residue (self); - g_free (self->residue); - self->residue = NULL; - } - self->kernel_length = val; - self->latency = val / 2; - lpwsinc_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); - lpwsinc_build_kernel (self); - GST_BASE_TRANSFORM_UNLOCK (self); - break; - case PROP_MODE: - GST_BASE_TRANSFORM_LOCK (self); - self->mode = g_value_get_enum (value); - lpwsinc_build_kernel (self); - GST_BASE_TRANSFORM_UNLOCK (self); - break; - case PROP_WINDOW: - GST_BASE_TRANSFORM_LOCK (self); - self->window = g_value_get_enum (value); - lpwsinc_build_kernel (self); - GST_BASE_TRANSFORM_UNLOCK (self); - break; - default: - G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); - break; - } -} - -static void -lpwsinc_get_property (GObject * object, guint prop_id, GValue * value, - GParamSpec * pspec) -{ - GstLPWSinc *self = GST_LPWSINC (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; - } -} diff --git a/gst/filter/gstlpwsinc.h b/gst/filter/gstlpwsinc.h deleted file mode 100644 index f56f5a4d..00000000 --- a/gst/filter/gstlpwsinc.h +++ /dev/null @@ -1,88 +0,0 @@ -/* -*- c-basic-offset: 2 -*- - * - * GStreamer - * Copyright (C) 1999-2001 Erik Walthinsen <omega@cse.ogi.edu> - * 2006 Dreamlab Technologies Ltd. <mathis.hofer@dreamlab.net> - * - * 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/ - * - */ - -#ifndef __GST_LPWSINC_H__ -#define __GST_LPWSINC_H__ - -#include "gstfilter.h" -#include <gst/gst.h> -#include <gst/audio/gstaudiofilter.h> - -G_BEGIN_DECLS - -#define GST_TYPE_LPWSINC \ - (gst_lpwsinc_get_type()) -#define GST_LPWSINC(obj) \ - (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_LPWSINC,GstLPWSinc)) -#define GST_LPWSINC_CLASS(klass) \ - (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_LPWSINC,GstLPWSincClass)) -#define GST_IS_LPWSINC(obj) \ - (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_LPWSINC)) -#define GST_IS_LPWSINC_CLASS(klass) \ - (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_LPWSINC)) - -typedef struct _GstLPWSinc GstLPWSinc; -typedef struct _GstLPWSincClass GstLPWSincClass; - -typedef void (*GstLPWSincProcessFunc) (GstLPWSinc *, guint8 *, guint8 *, guint); - -/** - * GstLPWSinc: - * - * Opaque data structure. - */ -struct _GstLPWSinc { - GstAudioFilter element; - - /* < private > */ - GstLPWSincProcessFunc 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; -}; - -struct _GstLPWSincClass { - GstAudioFilterClass parent_class; -}; - -GType gst_lpwsinc_get_type (void); - -G_END_DECLS - -#endif /* __GST_LPWSINC_H__ */ |