From: Sebastian Dröge Date: Fri, 10 Aug 2007 05:20:06 +0000 (+0000) Subject: gst/filter/gstbpwsinc.*: Apply the same changes to the bandpass filter: X-Git-Tag: 1.19.3~509^2~11794 X-Git-Url: http://review.tizen.org/git/?a=commitdiff_plain;h=86dab97c0258d1ae42e0fade45d22844b9466797;p=platform%2Fupstream%2Fgstreamer.git gst/filter/gstbpwsinc.*: Apply the same changes to the bandpass filter: Original commit message from CVS: * gst/filter/gstbpwsinc.c: (gst_bpwsinc_class_init), (gst_bpwsinc_init), (process_32), (process_64), (bpwsinc_build_kernel), (bpwsinc_setup), (bpwsinc_get_unit_size), (bpwsinc_transform), (bpwsinc_set_property), (bpwsinc_get_property): * gst/filter/gstbpwsinc.h: Apply the same changes to the bandpass filter: - Support double input - Fix processing for input with >1 channels - Specify frequency in Hz - Specify actual filter kernel length - Use transform instead of transform_ip as we're working out of place anyway - Factor out filter kernel generation and update the filter kernel when the properties are set Fix bandpass filter kernel generation to actually generate a bandpass filter by creating a highpass instead of a second lowpass. * gst/filter/gstlpwsinc.c: (gst_lpwsinc_class_init): Small formatting fix. --- diff --git a/gst/audiofx/audiowsincband.c b/gst/audiofx/audiowsincband.c index f2a4b60..15e35be 100644 --- a/gst/audiofx/audiowsincband.c +++ b/gst/audiofx/audiowsincband.c @@ -3,6 +3,7 @@ * 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 @@ -30,9 +31,6 @@ * is probably the bottleneck * - Implement a band reject mode (spectral inversion) * - Allow choosing between different windows (blackman, hanning, ...) - * - Specify filter length instead of 2*N+1 - * FIXME: - Doesn't work at all with >1 channels - * - Is bandreject, not bandpass */ #ifdef HAVE_CONFIG_H @@ -56,7 +54,8 @@ GST_ELEMENT_DETAILS ("Band-pass Windowed sinc filter", "Band-pass Windowed sinc filter", "Thomas , " "Steven W. Smith, " - "Dreamlab Technologies Ltd. "); + "Dreamlab Technologies Ltd. , " + "Sebastian Dröge "); /* Filter signals and args */ enum @@ -74,11 +73,11 @@ enum }; #define ALLOWED_CAPS \ - "audio/x-raw-float," \ - " width = (int) 32, " \ - " endianness = (int) BYTE_ORDER," \ - " rate = (int) [ 1, MAX ]," \ - " channels = (int) [ 1, MAX ]" + "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 Windowed sinc filter plugin"); @@ -91,8 +90,10 @@ static void bpwsinc_set_property (GObject * object, guint prop_id, static void bpwsinc_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec); -static GstFlowReturn bpwsinc_transform_ip (GstBaseTransform * base, - GstBuffer * outbuf); +static GstFlowReturn bpwsinc_transform (GstBaseTransform * base, + GstBuffer * inbuf, GstBuffer * outbuf); +static gboolean bpwsinc_get_unit_size (GstBaseTransform * base, GstCaps * caps, + guint * size); static gboolean bpwsinc_setup (GstAudioFilter * base, GstRingBufferSpec * format); @@ -145,133 +146,248 @@ gst_bpwsinc_class_init (GstBPWSincClass * klass) g_object_class_install_property (gobject_class, PROP_LOWER_FREQUENCY, g_param_spec_double ("lower-frequency", "Lower Frequency", - "Cut-off lower frequency (relative to sample rate)", - 0.0, 0.5, 0, G_PARAM_READWRITE)); + "Cut-off lower frequency (Hz)", + 0.0, G_MAXDOUBLE, 0, G_PARAM_READWRITE)); g_object_class_install_property (gobject_class, PROP_UPPER_FREQUENCY, g_param_spec_double ("upper-frequency", "Upper Frequency", - "Cut-off upper frequency (relative to sample rate)", - 0.0, 0.5, 0, G_PARAM_READWRITE)); + "Cut-off upper frequency (Hz)", + 0.0, G_MAXDOUBLE, 0, G_PARAM_READWRITE)); g_object_class_install_property (gobject_class, PROP_LENGTH, g_param_spec_int ("length", "Length", - "N such that the filter length = 2N + 1", - 1, G_MAXINT, 1, G_PARAM_READWRITE)); + "Filter kernel length, will be rounded to the next odd number", + 3, G_MAXINT, 101, G_PARAM_READWRITE)); - trans_class->transform_ip = GST_DEBUG_FUNCPTR (bpwsinc_transform_ip); + trans_class->transform = GST_DEBUG_FUNCPTR (bpwsinc_transform); + trans_class->get_unit_size = GST_DEBUG_FUNCPTR (bpwsinc_get_unit_size); GST_AUDIO_FILTER_CLASS (klass)->setup = GST_DEBUG_FUNCPTR (bpwsinc_setup); } static void gst_bpwsinc_init (GstBPWSinc * self, GstBPWSincClass * g_class) { - self->wing_size = 50; - self->lower_frequency = 0.25; - self->upper_frequency = 0.3; + self->kernel_length = 101; + self->lower_frequency = 0.0; + self->upper_frequency = 0.0; self->kernel = NULL; + self->have_kernel = FALSE; self->residue = NULL; } +static void +process_32 (GstBPWSinc * self, gfloat * src, gfloat * dst, guint input_samples) +{ + gint kernel_length = self->kernel_length; + gint i, j, k, l; + gint channels = GST_AUDIO_FILTER (self)->format.channels; -/* GstAudioFilter vmethod implementations */ + /* 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]; + } -/* get notified of caps and plug in the correct process function */ -static gboolean -bpwsinc_setup (GstAudioFilter * base, GstRingBufferSpec * format) + /* copy the tail of the current input buffer to the residue */ + for (i = 0; i < kernel_length * channels; i++) + self->residue[i] = src[input_samples - kernel_length * channels + i]; +} + +static void +process_64 (GstBPWSinc * self, gdouble * src, gdouble * dst, + guint input_samples) { - int i = 0; - double sum = 0.0; - int len = 0; - double *kernel_lp, *kernel_hp; - GstBPWSinc *self = GST_BPWSINC (base); + gint kernel_length = self->kernel_length; + gint i, j, k, l; + gint channels = GST_AUDIO_FILTER (self)->format.channels; - len = self->wing_size; - /* fill the lp kernel - * FIXME: refactor to own function, this is not caps related - */ + /* 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 */ + for (i = 0; i < kernel_length * channels; i++) + self->residue[i] = src[input_samples - kernel_length * channels + i]; +} + +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; + } + + /* fill the lp kernel */ GST_DEBUG ("bpwsinc: initializing LP kernel of length %d with cut-off %f", - len * 2 + 1, self->lower_frequency); - kernel_lp = (double *) g_malloc (sizeof (double) * (2 * len + 1)); - for (i = 0; i <= len * 2; ++i) { - if (i == len) - kernel_lp[i] = 2 * M_PI * self->lower_frequency; + len, self->lower_frequency); + + 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 (2 * M_PI * self->lower_frequency * (i - len)) - / (i - len); + kernel_lp[i] = sin (w * (i - len / 2)) + / (i - len / 2); /* Blackman windowing */ - kernel_lp[i] *= (0.42 - 0.5 * cos (M_PI * i / len) - + 0.08 * cos (2 * M_PI * i / len)); + 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 * 2; ++i) + for (i = 0; i < len; ++i) sum += kernel_lp[i]; - for (i = 0; i <= len * 2; ++i) + for (i = 0; i < len; ++i) kernel_lp[i] /= sum; /* fill the hp kernel */ GST_DEBUG ("bpwsinc: initializing HP kernel of length %d with cut-off %f", - len * 2 + 1, self->upper_frequency); - kernel_hp = (double *) g_malloc (sizeof (double) * (2 * len + 1)); - for (i = 0; i <= len * 2; ++i) { - if (i == len) - kernel_hp[i] = 2 * M_PI * self->upper_frequency; + len, self->upper_frequency); + + 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 (2 * M_PI * self->upper_frequency * (i - len)) - / (i - len); + kernel_hp[i] = sin (w * (i - len / 2)) + / (i - len / 2); /* Blackman windowing */ - kernel_hp[i] *= (0.42 - 0.5 * cos (M_PI * i / len) - + 0.08 * cos (2 * M_PI * i / len)); + 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 * 2; ++i) + for (i = 0; i < len; ++i) sum += kernel_hp[i]; - for (i = 0; i <= len * 2; ++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 = (double *) g_malloc (sizeof (double) * (2 * len + 1)); + self->kernel = g_new (gdouble, len); - for (i = 0; i <= len * 2; ++i) + for (i = 0; i < len; ++i) self->kernel[i] = kernel_lp[i] + kernel_hp[i]; - /* do spectral inversion to go from band reject to bandpass */ - for (i = 0; i <= len * 2; ++i) - self->kernel[i] = -self->kernel[i]; - self->kernel[len] += 1; - /* free the helper kernels */ g_free (kernel_lp); g_free (kernel_hp); + /* do spectral inversion to go from bandreject to bandpass */ + 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) g_free (self->residue); - self->residue = (gfloat *) g_malloc (sizeof (gfloat) * (len * 2 + 1)); - for (i = 0; i <= len * 2; ++i) - self->residue[i] = 0.0; + self->residue = g_new0 (gdouble, len); +} + +/* 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 gboolean +bpwsinc_get_unit_size (GstBaseTransform * base, GstCaps * caps, guint * size) +{ + gint width, channels; + GstStructure *structure; + gboolean ret; + + g_assert (size); + + structure = gst_caps_get_structure (caps, 0); + ret = gst_structure_get_int (structure, "width", &width); + ret &= gst_structure_get_int (structure, "channels", &channels); + + *size = width * channels / 8; + + return ret; +} + static GstFlowReturn -bpwsinc_transform_ip (GstBaseTransform * base, GstBuffer * outbuf) +bpwsinc_transform (GstBaseTransform * base, GstBuffer * inbuf, + GstBuffer * outbuf) { GstBPWSinc *self = GST_BPWSINC (base); GstClockTime timestamp; - - gfloat *src; - gfloat *input; - int residue_samples; - gint input_samples; - gint total_samples; - int i, j; + gint input_samples = + GST_BUFFER_SIZE (outbuf) / (GST_AUDIO_FILTER (self)->format.width / 8); /* don't process data in passthrough-mode */ if (gst_base_transform_is_passthrough (base)) @@ -283,38 +399,11 @@ bpwsinc_transform_ip (GstBaseTransform * base, GstBuffer * outbuf) if (GST_CLOCK_TIME_IS_VALID (timestamp)) gst_object_sync_values (G_OBJECT (self), timestamp); - /* FIXME: out of laziness, we copy the left-over bit from last buffer - * together with the incoming buffer to a new buffer to make the loop - * easy; self could be a lot more optimized though - * to make amends we keep the incoming buffer around and write our - * output samples there */ - - src = (gfloat *) GST_BUFFER_DATA (outbuf); - residue_samples = self->wing_size * 2 + 1; - input_samples = GST_BUFFER_SIZE (outbuf) / sizeof (gfloat); - total_samples = residue_samples + input_samples; + if (!self->have_kernel) + bpwsinc_build_kernel (self); - input = (gfloat *) g_malloc (sizeof (gfloat) * total_samples); - - /* copy the left-over bit */ - memcpy (input, self->residue, sizeof (gfloat) * residue_samples); - - /* copy the new buffer */ - memcpy (&input[residue_samples], src, sizeof (gfloat) * input_samples); - /* copy the tail of the current input buffer to the residue */ - memcpy (self->residue, &src[input_samples - residue_samples], - sizeof (gfloat) * residue_samples); - - /* convolution */ - /* since we copied the previous set of samples we needed before the actual - * input data, we need to add the filter length to our indices for input */ - for (i = 0; i < input_samples; ++i) { - src[i] = 0.0; - for (j = 0; j < residue_samples; ++j) - src[i] += input[i - j + residue_samples] * self->kernel[j]; - } - - g_free (input); + self->process (self, GST_BUFFER_DATA (inbuf), GST_BUFFER_DATA (outbuf), + input_samples); return GST_FLOW_OK; } @@ -328,14 +417,29 @@ bpwsinc_set_property (GObject * object, guint prop_id, const GValue * value, g_return_if_fail (GST_IS_BPWSINC (self)); switch (prop_id) { - case PROP_LENGTH: - self->wing_size = g_value_get_int (value); + case PROP_LENGTH:{ + gint val; + + GST_BASE_TRANSFORM_LOCK (self); + val = g_value_get_int (value); + if (val % 2 == 0) + val++; + self->kernel_length = val; + bpwsinc_build_kernel (self); + GST_BASE_TRANSFORM_UNLOCK (self); break; + } case PROP_LOWER_FREQUENCY: + GST_BASE_TRANSFORM_LOCK (self); self->lower_frequency = g_value_get_double (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_double (value); + bpwsinc_build_kernel (self); + GST_BASE_TRANSFORM_UNLOCK (self); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); @@ -351,7 +455,7 @@ bpwsinc_get_property (GObject * object, guint prop_id, GValue * value, switch (prop_id) { case PROP_LENGTH: - g_value_set_int (value, self->wing_size); + g_value_set_int (value, self->kernel_length); break; case PROP_LOWER_FREQUENCY: g_value_set_double (value, self->lower_frequency); diff --git a/gst/audiofx/audiowsincband.h b/gst/audiofx/audiowsincband.h index af5938d..3bbb3e0 100644 --- a/gst/audiofx/audiowsincband.h +++ b/gst/audiofx/audiowsincband.h @@ -25,10 +25,6 @@ * chapter 16 * available at http://www.dspguide.com/ * - * FIXME: - * - this filter is totally unoptimized ! - * - we do not destroy the allocated memory for filters and residue - * - this might be improved upon with bytestream */ #ifndef __GST_BPWSINC_H__ @@ -54,6 +50,8 @@ G_BEGIN_DECLS typedef struct _GstBPWSinc GstBPWSinc; typedef struct _GstBPWSincClass GstBPWSincClass; +typedef void (*GstBPWSincProcessFunc) (GstBPWSinc *, guint8 *, guint8 *, guint); + /** * GstBPWSinc: * @@ -62,13 +60,15 @@ typedef struct _GstBPWSincClass GstBPWSincClass; struct _GstBPWSinc { GstAudioFilter element; - double frequency; - double lower_frequency, upper_frequency; - int wing_size; /* length of a "wing" of the filter; - actual length is 2 * wing_size + 1 */ + GstBPWSincProcessFunc process; + + gdouble frequency; + gdouble lower_frequency, upper_frequency; + gint kernel_length; /* length of the filter kernel */ - gfloat *residue; /* buffer for left-over samples from previous buffer */ - double *kernel; + gdouble *residue; /* buffer for left-over samples from previous buffer */ + gdouble *kernel; + gboolean have_kernel; }; struct _GstBPWSincClass { diff --git a/gst/audiofx/audiowsinclimit.c b/gst/audiofx/audiowsinclimit.c index 24906d1..2debfe7 100644 --- a/gst/audiofx/audiowsinclimit.c +++ b/gst/audiofx/audiowsinclimit.c @@ -126,10 +126,10 @@ gst_lpwsinc_window_get_type (void) #define ALLOWED_CAPS \ - "audio/x-raw-float," \ + "audio/x-raw-float, " \ " width = (int) { 32, 64 }, " \ - " endianness = (int) BYTE_ORDER," \ - " rate = (int) [ 1, MAX ]," \ + " endianness = (int) BYTE_ORDER, " \ + " rate = (int) [ 1, MAX ], " \ " channels = (int) [ 1, MAX ]" #define DEBUG_INIT(bla) \ @@ -199,7 +199,7 @@ gst_lpwsinc_class_init (GstLPWSincClass * klass) g_object_class_install_property (gobject_class, PROP_FREQUENCY, g_param_spec_double ("frequency", "Frequency", - "Cut-off Frequency", 0.0, G_MAXDOUBLE, 0.0, + "Cut-off Frequency (Hz)", 0.0, G_MAXDOUBLE, 0.0, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE)); g_object_class_install_property (gobject_class, PROP_LENGTH,