diff options
Diffstat (limited to 'gst/audiofx/audiowsincband.c')
-rw-r--r-- | gst/audiofx/audiowsincband.c | 570 |
1 files changed, 70 insertions, 500 deletions
diff --git a/gst/audiofx/audiowsincband.c b/gst/audiofx/audiowsincband.c index 60848b1b..62034308 100644 --- a/gst/audiofx/audiowsincband.c +++ b/gst/audiofx/audiowsincband.c @@ -3,7 +3,7 @@ * GStreamer * Copyright (C) 1999-2001 Erik Walthinsen <omega@cse.ogi.edu> * 2006 Dreamlab Technologies Ltd. <mathis.hofer@dreamlab.net> - * 2007 Sebastian Dröge <slomo@circular-chaos.org> + * 2007-2009 Sebastian Dröge <sebastian.droege@collabora.co.uk> * * This library is free software; you can redistribute it and/or * modify it under the terms of the GNU Library General Public @@ -74,25 +74,9 @@ #include "audiowsincband.h" -#define GST_CAT_DEFAULT gst_audio_wsincband_debug +#define GST_CAT_DEFAULT gst_gst_audio_wsincband_debug GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT); -static const GstElementDetails audio_wsincband_details = -GST_ELEMENT_DETAILS ("Band pass & band reject filter", - "Filter/Effect/Audio", - "Band pass and band reject windowed sinc filter", - "Thomas Vander Stichele <thomas at apestaart dot org>, " - "Steven W. Smith, " - "Dreamlab Technologies Ltd. <mathis.hofer@dreamlab.net>, " - "Sebastian Dröge <slomo@circular-chaos.org>"); - -/* Filter signals and args */ -enum -{ - /* FILL ME */ - LAST_SIGNAL -}; - enum { PROP_0, @@ -109,9 +93,9 @@ enum MODE_BAND_REJECT }; -#define GST_TYPE_AUDIO_WSINC_BAND_MODE (gst_audio_wsincband_mode_get_type ()) +#define GST_TYPE_AUDIO_WSINC_BAND_MODE (gst_gst_audio_wsincband_mode_get_type ()) static GType -gst_audio_wsincband_mode_get_type (void) +gst_gst_audio_wsincband_mode_get_type (void) { static GType gtype = 0; @@ -135,9 +119,9 @@ enum WINDOW_BLACKMAN }; -#define GST_TYPE_AUDIO_WSINC_BAND_WINDOW (gst_audio_wsincband_window_get_type ()) +#define GST_TYPE_AUDIO_WSINC_BAND_WINDOW (gst_gst_audio_wsincband_window_get_type ()) static GType -gst_audio_wsincband_window_get_type (void) +gst_gst_audio_wsincband_window_get_type (void) { static GType gtype = 0; @@ -155,193 +139,96 @@ gst_audio_wsincband_window_get_type (void) return gtype; } -#define ALLOWED_CAPS \ - "audio/x-raw-float, " \ - " width = (int) { 32, 64 }, " \ - " endianness = (int) BYTE_ORDER, " \ - " rate = (int) [ 1, MAX ], " \ - " channels = (int) [ 1, MAX ] " - #define DEBUG_INIT(bla) \ - GST_DEBUG_CATEGORY_INIT (gst_audio_wsincband_debug, "audiowsincband", 0, \ + GST_DEBUG_CATEGORY_INIT (gst_gst_audio_wsincband_debug, "audiowsincband", 0, \ "Band-pass and Band-reject Windowed sinc filter plugin"); -GST_BOILERPLATE_FULL (GstAudioWSincBand, audio_wsincband, GstAudioFilter, - GST_TYPE_AUDIO_FILTER, DEBUG_INIT); +GST_BOILERPLATE_FULL (GstAudioWSincBand, gst_audio_wsincband, GstAudioFilter, + GST_TYPE_AUDIO_FX_BASE_FIR_FILTER, DEBUG_INIT); -static void audio_wsincband_set_property (GObject * object, guint prop_id, +static void gst_audio_wsincband_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec); -static void audio_wsincband_get_property (GObject * object, guint prop_id, +static void gst_audio_wsincband_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec); -static GstFlowReturn audio_wsincband_transform (GstBaseTransform * base, - GstBuffer * inbuf, GstBuffer * outbuf); -static gboolean audio_wsincband_start (GstBaseTransform * base); -static gboolean audio_wsincband_event (GstBaseTransform * base, - GstEvent * event); - -static gboolean audio_wsincband_setup (GstAudioFilter * base, +static gboolean gst_audio_wsincband_setup (GstAudioFilter * base, GstRingBufferSpec * format); -static gboolean audio_wsincband_query (GstPad * pad, GstQuery * query); -static const GstQueryType *audio_wsincband_query_type (GstPad * pad); - /* Element class */ - static void -audio_wsincband_dispose (GObject * object) -{ - GstAudioWSincBand *self = GST_AUDIO_WSINC_BAND (object); - - if (self->residue) { - g_free (self->residue); - self->residue = NULL; - } - - if (self->kernel) { - g_free (self->kernel); - self->kernel = NULL; - } - - G_OBJECT_CLASS (parent_class)->dispose (object); -} - -static void -audio_wsincband_base_init (gpointer g_class) +gst_audio_wsincband_base_init (gpointer g_class) { GstElementClass *element_class = GST_ELEMENT_CLASS (g_class); - GstCaps *caps; - gst_element_class_set_details (element_class, &audio_wsincband_details); - - caps = gst_caps_from_string (ALLOWED_CAPS); - gst_audio_filter_class_add_pad_templates (GST_AUDIO_FILTER_CLASS (g_class), - caps); - gst_caps_unref (caps); + gst_element_class_set_details_simple (element_class, + "Band pass & band reject filter", "Filter/Effect/Audio", + "Band pass and band reject windowed sinc filter", + "Thomas Vander Stichele <thomas at apestaart dot org>, " + "Steven W. Smith, " + "Dreamlab Technologies Ltd. <mathis.hofer@dreamlab.net>, " + "Sebastian Dröge <sebastian.droege@collabora.co.uk>"); } static void -audio_wsincband_class_init (GstAudioWSincBandClass * klass) +gst_audio_wsincband_class_init (GstAudioWSincBandClass * klass) { - GObjectClass *gobject_class; - GstBaseTransformClass *trans_class; - GstAudioFilterClass *filter_class; - - gobject_class = (GObjectClass *) klass; - trans_class = (GstBaseTransformClass *) klass; - filter_class = (GstAudioFilterClass *) klass; + GObjectClass *gobject_class = (GObjectClass *) klass; + GstAudioFilterClass *filter_class = (GstAudioFilterClass *) klass; - gobject_class->set_property = audio_wsincband_set_property; - gobject_class->get_property = audio_wsincband_get_property; - gobject_class->dispose = audio_wsincband_dispose; + gobject_class->set_property = gst_audio_wsincband_set_property; + gobject_class->get_property = gst_audio_wsincband_get_property; /* FIXME: Don't use the complete possible range but restrict the upper boundary * so automatically generated UIs can use a slider */ g_object_class_install_property (gobject_class, PROP_LOWER_FREQUENCY, g_param_spec_float ("lower-frequency", "Lower Frequency", - "Cut-off lower frequency (Hz)", 0.0, 100000.0, 0, G_PARAM_READWRITE)); + "Cut-off lower frequency (Hz)", 0.0, 100000.0, 0, + G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (gobject_class, PROP_UPPER_FREQUENCY, g_param_spec_float ("upper-frequency", "Upper Frequency", - "Cut-off upper frequency (Hz)", 0.0, 100000.0, 0, G_PARAM_READWRITE)); + "Cut-off upper frequency (Hz)", 0.0, 100000.0, 0, + G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (gobject_class, PROP_LENGTH, g_param_spec_int ("length", "Length", - "Filter kernel length, will be rounded to the next odd number", - 3, 50000, 101, G_PARAM_READWRITE)); + "Filter kernel length, will be rounded to the next odd number", 3, + 50000, 101, + G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (gobject_class, PROP_MODE, g_param_spec_enum ("mode", "Mode", "Band pass or band reject mode", GST_TYPE_AUDIO_WSINC_BAND_MODE, - MODE_BAND_PASS, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE)); + MODE_BAND_PASS, + G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (gobject_class, PROP_WINDOW, g_param_spec_enum ("window", "Window", "Window function to use", GST_TYPE_AUDIO_WSINC_BAND_WINDOW, - WINDOW_HAMMING, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE)); + WINDOW_HAMMING, + G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE | G_PARAM_STATIC_STRINGS)); - trans_class->transform = GST_DEBUG_FUNCPTR (audio_wsincband_transform); - trans_class->start = GST_DEBUG_FUNCPTR (audio_wsincband_start); - trans_class->event = GST_DEBUG_FUNCPTR (audio_wsincband_event); - filter_class->setup = GST_DEBUG_FUNCPTR (audio_wsincband_setup); + filter_class->setup = GST_DEBUG_FUNCPTR (gst_audio_wsincband_setup); } static void -audio_wsincband_init (GstAudioWSincBand * self, +gst_audio_wsincband_init (GstAudioWSincBand * self, GstAudioWSincBandClass * g_class) { self->kernel_length = 101; - self->latency = 50; self->lower_frequency = 0.0; self->upper_frequency = 0.0; self->mode = MODE_BAND_PASS; self->window = WINDOW_HAMMING; - self->kernel = NULL; - self->have_kernel = FALSE; - self->residue = NULL; - - self->residue_length = 0; - self->next_ts = GST_CLOCK_TIME_NONE; - self->next_off = GST_BUFFER_OFFSET_NONE; - - gst_pad_set_query_function (GST_BASE_TRANSFORM (self)->srcpad, - audio_wsincband_query); - gst_pad_set_query_type_function (GST_BASE_TRANSFORM (self)->srcpad, - audio_wsincband_query_type); -} - -#define DEFINE_PROCESS_FUNC(width,ctype) \ -static void \ -process_##width (GstAudioWSincBand * self, g##ctype * src, g##ctype * dst, guint input_samples) \ -{ \ - gint kernel_length = self->kernel_length; \ - gint i, j, k, l; \ - gint channels = GST_AUDIO_FILTER (self)->format.channels; \ - gint res_start; \ - \ - /* convolution */ \ - for (i = 0; i < input_samples; i++) { \ - dst[i] = 0.0; \ - k = i % channels; \ - l = i / channels; \ - for (j = 0; j < kernel_length; j++) \ - if (l < j) \ - dst[i] += \ - self->residue[(kernel_length + l - j) * channels + \ - k] * self->kernel[j]; \ - else \ - dst[i] += src[(l - j) * channels + k] * self->kernel[j]; \ - } \ - \ - /* copy the tail of the current input buffer to the residue, while \ - * keeping parts of the residue if the input buffer is smaller than \ - * the kernel length */ \ - if (input_samples < kernel_length * channels) \ - res_start = kernel_length * channels - input_samples; \ - else \ - res_start = 0; \ - \ - for (i = 0; i < res_start; i++) \ - self->residue[i] = self->residue[i + input_samples]; \ - for (i = res_start; i < kernel_length * channels; i++) \ - self->residue[i] = src[input_samples - kernel_length * channels + i]; \ - \ - self->residue_length += kernel_length * channels - res_start; \ - if (self->residue_length > kernel_length * channels) \ - self->residue_length = kernel_length * channels; \ } -DEFINE_PROCESS_FUNC (32, float); -DEFINE_PROCESS_FUNC (64, double); - -#undef DEFINE_PROCESS_FUNC - static void -audio_wsincband_build_kernel (GstAudioWSincBand * self) +gst_audio_wsincband_build_kernel (GstAudioWSincBand * self) { gint i = 0; gdouble sum = 0.0; gint len = 0; gdouble *kernel_lp, *kernel_hp; gdouble w; + gdouble *kernel; len = self->kernel_length; @@ -369,7 +256,7 @@ audio_wsincband_build_kernel (GstAudioWSincBand * self) self->upper_frequency = tmp; } - GST_DEBUG ("audio_wsincband: initializing filter kernel of length %d " + GST_DEBUG ("gst_audio_wsincband: initializing filter kernel of length %d " "with lower frequency %.2lf Hz " ", upper frequency %.2lf Hz for mode %s", len, self->lower_frequency, self->upper_frequency, @@ -431,12 +318,10 @@ audio_wsincband_build_kernel (GstAudioWSincBand * self) kernel_hp[len / 2] += 1; /* combine the two kernels */ - if (self->kernel) - g_free (self->kernel); - self->kernel = g_new (gdouble, len); + kernel = g_new (gdouble, len); for (i = 0; i < len; ++i) - self->kernel[i] = kernel_lp[i] + kernel_hp[i]; + kernel[i] = kernel_lp[i] + kernel_hp[i]; /* free the helper kernels */ g_free (kernel_lp); @@ -446,338 +331,29 @@ audio_wsincband_build_kernel (GstAudioWSincBand * self) * if specified */ if (self->mode == MODE_BAND_PASS) { for (i = 0; i < len; ++i) - self->kernel[i] = -self->kernel[i]; - self->kernel[len / 2] += 1; - } - - /* set up the residue memory space */ - if (!self->residue) { - self->residue = - g_new0 (gdouble, len * GST_AUDIO_FILTER (self)->format.channels); - self->residue_length = 0; - } - - self->have_kernel = TRUE; -} - -static void -audio_wsincband_push_residue (GstAudioWSincBand * self) -{ - GstBuffer *outbuf; - GstFlowReturn res; - gint rate = GST_AUDIO_FILTER (self)->format.rate; - gint channels = GST_AUDIO_FILTER (self)->format.channels; - gint outsize, outsamples; - gint diffsize, diffsamples; - guint8 *in, *out; - - /* Calculate the number of samples and their memory size that - * should be pushed from the residue */ - outsamples = MIN (self->latency, self->residue_length / channels); - outsize = outsamples * channels * (GST_AUDIO_FILTER (self)->format.width / 8); - if (outsize == 0) - return; - - /* Process the difference between latency and residue_length samples - * to start at the actual data instead of starting at the zeros before - * when we only got one buffer smaller than latency */ - diffsamples = self->latency - self->residue_length / channels; - diffsize = - diffsamples * channels * (GST_AUDIO_FILTER (self)->format.width / 8); - if (diffsize > 0) { - in = g_new0 (guint8, diffsize); - out = g_new0 (guint8, diffsize); - self->process (self, in, out, diffsamples * channels); - g_free (in); - g_free (out); - } - - res = gst_pad_alloc_buffer (GST_BASE_TRANSFORM (self)->srcpad, - GST_BUFFER_OFFSET_NONE, outsize, - GST_PAD_CAPS (GST_BASE_TRANSFORM (self)->srcpad), &outbuf); - - if (G_UNLIKELY (res != GST_FLOW_OK)) { - GST_WARNING_OBJECT (self, "failed allocating buffer of %d bytes", outsize); - return; - } - - /* Convolve the residue with zeros to get the actual remaining data */ - in = g_new0 (guint8, outsize); - self->process (self, in, GST_BUFFER_DATA (outbuf), outsamples * channels); - g_free (in); - - /* Set timestamp, offset, etc from the values we - * saved when processing the regular buffers */ - if (GST_CLOCK_TIME_IS_VALID (self->next_ts)) - GST_BUFFER_TIMESTAMP (outbuf) = self->next_ts; - else - GST_BUFFER_TIMESTAMP (outbuf) = 0; - GST_BUFFER_DURATION (outbuf) = - gst_util_uint64_scale (outsamples, GST_SECOND, rate); - self->next_ts += gst_util_uint64_scale (outsamples, GST_SECOND, rate); - - if (self->next_off != GST_BUFFER_OFFSET_NONE) { - GST_BUFFER_OFFSET (outbuf) = self->next_off; - GST_BUFFER_OFFSET_END (outbuf) = self->next_off + outsamples; - } - - GST_DEBUG_OBJECT (self, "Pushing residue buffer of size %d with timestamp: %" - GST_TIME_FORMAT ", duration: %" GST_TIME_FORMAT ", offset: %lld," - " offset_end: %lld, nsamples: %d", GST_BUFFER_SIZE (outbuf), - GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf)), - GST_TIME_ARGS (GST_BUFFER_DURATION (outbuf)), GST_BUFFER_OFFSET (outbuf), - GST_BUFFER_OFFSET_END (outbuf), outsamples); - - res = gst_pad_push (GST_BASE_TRANSFORM (self)->srcpad, outbuf); - - if (G_UNLIKELY (res != GST_FLOW_OK)) { - GST_WARNING_OBJECT (self, "failed to push residue"); + kernel[i] = -kernel[i]; + kernel[len / 2] += 1; } + gst_audio_fx_base_fir_filter_set_kernel (GST_AUDIO_FX_BASE_FIR_FILTER (self), + kernel, self->kernel_length, (len - 1) / 2); } /* GstAudioFilter vmethod implementations */ /* get notified of caps and plug in the correct process function */ static gboolean -audio_wsincband_setup (GstAudioFilter * base, GstRingBufferSpec * format) -{ - GstAudioWSincBand *self = GST_AUDIO_WSINC_BAND (base); - - gboolean ret = TRUE; - - if (format->width == 32) - self->process = (GstAudioWSincBandProcessFunc) process_32; - else if (format->width == 64) - self->process = (GstAudioWSincBandProcessFunc) process_64; - else - ret = FALSE; - - self->have_kernel = FALSE; - - return TRUE; -} - -/* GstBaseTransform vmethod implementations */ - -static GstFlowReturn -audio_wsincband_transform (GstBaseTransform * base, GstBuffer * inbuf, - GstBuffer * outbuf) -{ - GstAudioWSincBand *self = GST_AUDIO_WSINC_BAND (base); - GstClockTime timestamp; - gint channels = GST_AUDIO_FILTER (self)->format.channels; - gint rate = GST_AUDIO_FILTER (self)->format.rate; - gint input_samples = - GST_BUFFER_SIZE (outbuf) / (GST_AUDIO_FILTER (self)->format.width / 8); - gint output_samples = input_samples; - gint diff; - - /* FIXME: subdivide GST_BUFFER_SIZE into small chunks for smooth fades */ - timestamp = GST_BUFFER_TIMESTAMP (outbuf); - if (GST_CLOCK_TIME_IS_VALID (timestamp)) - gst_object_sync_values (G_OBJECT (self), timestamp); - - if (!self->have_kernel) - audio_wsincband_build_kernel (self); - - /* Reset the residue if already existing on discont buffers */ - if (GST_BUFFER_IS_DISCONT (inbuf)) { - if (channels && self->residue) - memset (self->residue, 0, channels * - self->kernel_length * sizeof (gdouble)); - self->residue_length = 0; - self->next_ts = GST_CLOCK_TIME_NONE; - self->next_off = GST_BUFFER_OFFSET_NONE; - } - - /* Calculate the number of samples we can push out now without outputting - * kernel_length/2 zeros in the beginning */ - diff = (self->kernel_length / 2) * channels - self->residue_length; - if (diff > 0) - output_samples -= diff; - - self->process (self, GST_BUFFER_DATA (inbuf), GST_BUFFER_DATA (outbuf), - input_samples); - - if (output_samples <= 0) { - /* Drop buffer and save original timestamp/offset for later use */ - if (!GST_CLOCK_TIME_IS_VALID (self->next_ts) - && GST_BUFFER_TIMESTAMP_IS_VALID (outbuf)) - self->next_ts = GST_BUFFER_TIMESTAMP (outbuf); - if (self->next_off == GST_BUFFER_OFFSET_NONE - && GST_BUFFER_OFFSET_IS_VALID (outbuf)) - self->next_off = GST_BUFFER_OFFSET (outbuf); - return GST_BASE_TRANSFORM_FLOW_DROPPED; - } else if (output_samples < input_samples) { - /* First (probably partial) buffer after starting from - * a clean residue. Use stored timestamp/offset here */ - if (GST_CLOCK_TIME_IS_VALID (self->next_ts)) - GST_BUFFER_TIMESTAMP (outbuf) = self->next_ts; - - if (self->next_off != GST_BUFFER_OFFSET_NONE) { - GST_BUFFER_OFFSET (outbuf) = self->next_off; - if (GST_BUFFER_OFFSET_END_IS_VALID (outbuf)) - GST_BUFFER_OFFSET_END (outbuf) = - self->next_off + output_samples / channels; - } else { - /* We dropped no buffer, offset is valid, offset_end must be adjusted by diff */ - if (GST_BUFFER_OFFSET_END_IS_VALID (outbuf)) - GST_BUFFER_OFFSET_END (outbuf) -= diff / channels; - } - - if (GST_BUFFER_DURATION_IS_VALID (outbuf)) - GST_BUFFER_DURATION (outbuf) -= - gst_util_uint64_scale (diff, GST_SECOND, channels * rate); - - GST_BUFFER_DATA (outbuf) += - diff * (GST_AUDIO_FILTER (self)->format.width / 8); - GST_BUFFER_SIZE (outbuf) -= - diff * (GST_AUDIO_FILTER (self)->format.width / 8); - } else { - GstClockTime ts_latency = - gst_util_uint64_scale (self->latency, GST_SECOND, rate); - - /* Normal buffer, adjust timestamp/offset/etc by latency */ - if (GST_BUFFER_TIMESTAMP (outbuf) < ts_latency) { - GST_WARNING_OBJECT (self, "GST_BUFFER_TIMESTAMP (outbuf) < latency"); - GST_BUFFER_TIMESTAMP (outbuf) = 0; - } else { - GST_BUFFER_TIMESTAMP (outbuf) -= ts_latency; - } - - if (GST_BUFFER_OFFSET_IS_VALID (outbuf)) { - if (GST_BUFFER_OFFSET (outbuf) > self->latency) { - GST_BUFFER_OFFSET (outbuf) -= self->latency; - } else { - GST_WARNING_OBJECT (self, "GST_BUFFER_OFFSET (outbuf) < latency"); - GST_BUFFER_OFFSET (outbuf) = 0; - } - } - - if (GST_BUFFER_OFFSET_END_IS_VALID (outbuf)) { - if (GST_BUFFER_OFFSET_END (outbuf) > self->latency) { - GST_BUFFER_OFFSET_END (outbuf) -= self->latency; - } else { - GST_WARNING_OBJECT (self, "GST_BUFFER_OFFSET_END (outbuf) < latency"); - GST_BUFFER_OFFSET_END (outbuf) = 0; - } - } - } - - GST_DEBUG_OBJECT (self, "Pushing buffer of size %d with timestamp: %" - GST_TIME_FORMAT ", duration: %" GST_TIME_FORMAT ", offset: %lld," - " offset_end: %lld, nsamples: %d", GST_BUFFER_SIZE (outbuf), - GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf)), - GST_TIME_ARGS (GST_BUFFER_DURATION (outbuf)), GST_BUFFER_OFFSET (outbuf), - GST_BUFFER_OFFSET_END (outbuf), output_samples / channels); - - self->next_ts = GST_BUFFER_TIMESTAMP (outbuf) + GST_BUFFER_DURATION (outbuf); - self->next_off = GST_BUFFER_OFFSET_END (outbuf); - - return GST_FLOW_OK; -} - -static gboolean -audio_wsincband_start (GstBaseTransform * base) -{ - GstAudioWSincBand *self = GST_AUDIO_WSINC_BAND (base); - gint channels = GST_AUDIO_FILTER (self)->format.channels; - - /* Reset the residue if already existing */ - if (channels && self->residue) - memset (self->residue, 0, channels * - self->kernel_length * sizeof (gdouble)); - - self->residue_length = 0; - self->next_ts = GST_CLOCK_TIME_NONE; - self->next_off = GST_BUFFER_OFFSET_NONE; - - return TRUE; -} - -static gboolean -audio_wsincband_query (GstPad * pad, GstQuery * query) -{ - GstAudioWSincBand *self = GST_AUDIO_WSINC_BAND (gst_pad_get_parent (pad)); - gboolean res = TRUE; - - switch (GST_QUERY_TYPE (query)) { - case GST_QUERY_LATENCY: - { - GstClockTime min, max; - gboolean live; - guint64 latency; - GstPad *peer; - gint rate = GST_AUDIO_FILTER (self)->format.rate; - - if ((peer = gst_pad_get_peer (GST_BASE_TRANSFORM (self)->sinkpad))) { - if ((res = gst_pad_query (peer, query))) { - gst_query_parse_latency (query, &live, &min, &max); - - GST_DEBUG_OBJECT (self, "Peer latency: min %" - GST_TIME_FORMAT " max %" GST_TIME_FORMAT, - GST_TIME_ARGS (min), GST_TIME_ARGS (max)); - - /* add our own latency */ - latency = - (rate != 0) ? gst_util_uint64_scale (self->latency, GST_SECOND, - rate) : 0; - - GST_DEBUG_OBJECT (self, "Our latency: %" - GST_TIME_FORMAT, GST_TIME_ARGS (latency)); - - min += latency; - if (max != GST_CLOCK_TIME_NONE) - max += latency; - - GST_DEBUG_OBJECT (self, "Calculated total latency : min %" - GST_TIME_FORMAT " max %" GST_TIME_FORMAT, - GST_TIME_ARGS (min), GST_TIME_ARGS (max)); - - gst_query_set_latency (query, live, min, max); - } - gst_object_unref (peer); - } - break; - } - default: - res = gst_pad_query_default (pad, query); - break; - } - gst_object_unref (self); - return res; -} - -static const GstQueryType * -audio_wsincband_query_type (GstPad * pad) -{ - static const GstQueryType types[] = { - GST_QUERY_LATENCY, - 0 - }; - - return types; -} - -static gboolean -audio_wsincband_event (GstBaseTransform * base, GstEvent * event) +gst_audio_wsincband_setup (GstAudioFilter * base, GstRingBufferSpec * format) { GstAudioWSincBand *self = GST_AUDIO_WSINC_BAND (base); - switch (GST_EVENT_TYPE (event)) { - case GST_EVENT_EOS: - audio_wsincband_push_residue (self); - break; - default: - break; - } + gst_audio_wsincband_build_kernel (self); - return GST_BASE_TRANSFORM_CLASS (parent_class)->event (base, event); + return GST_AUDIO_FILTER_CLASS (parent_class)->setup (base, format); } static void -audio_wsincband_set_property (GObject * object, guint prop_id, +gst_audio_wsincband_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec) { GstAudioWSincBand *self = GST_AUDIO_WSINC_BAND (object); @@ -788,49 +364,43 @@ audio_wsincband_set_property (GObject * object, guint prop_id, case PROP_LENGTH:{ gint val; - GST_BASE_TRANSFORM_LOCK (self); + GST_OBJECT_LOCK (self); val = g_value_get_int (value); if (val % 2 == 0) val++; if (val != self->kernel_length) { - if (self->residue) { - audio_wsincband_push_residue (self); - g_free (self->residue); - self->residue = NULL; - } + gst_audio_fx_base_fir_filter_push_residue (GST_AUDIO_FX_BASE_FIR_FILTER + (self)); self->kernel_length = val; - self->latency = val / 2; - audio_wsincband_build_kernel (self); - gst_element_post_message (GST_ELEMENT (self), - gst_message_new_latency (GST_OBJECT (self))); + gst_audio_wsincband_build_kernel (self); } - GST_BASE_TRANSFORM_UNLOCK (self); + GST_OBJECT_UNLOCK (self); break; } case PROP_LOWER_FREQUENCY: - GST_BASE_TRANSFORM_LOCK (self); + GST_OBJECT_LOCK (self); self->lower_frequency = g_value_get_float (value); - audio_wsincband_build_kernel (self); - GST_BASE_TRANSFORM_UNLOCK (self); + gst_audio_wsincband_build_kernel (self); + GST_OBJECT_UNLOCK (self); break; case PROP_UPPER_FREQUENCY: - GST_BASE_TRANSFORM_LOCK (self); + GST_OBJECT_LOCK (self); self->upper_frequency = g_value_get_float (value); - audio_wsincband_build_kernel (self); - GST_BASE_TRANSFORM_UNLOCK (self); + gst_audio_wsincband_build_kernel (self); + GST_OBJECT_UNLOCK (self); break; case PROP_MODE: - GST_BASE_TRANSFORM_LOCK (self); + GST_OBJECT_LOCK (self); self->mode = g_value_get_enum (value); - audio_wsincband_build_kernel (self); - GST_BASE_TRANSFORM_UNLOCK (self); + gst_audio_wsincband_build_kernel (self); + GST_OBJECT_UNLOCK (self); break; case PROP_WINDOW: - GST_BASE_TRANSFORM_LOCK (self); + GST_OBJECT_LOCK (self); self->window = g_value_get_enum (value); - audio_wsincband_build_kernel (self); - GST_BASE_TRANSFORM_UNLOCK (self); + gst_audio_wsincband_build_kernel (self); + GST_OBJECT_UNLOCK (self); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); @@ -839,8 +409,8 @@ audio_wsincband_set_property (GObject * object, guint prop_id, } static void -audio_wsincband_get_property (GObject * object, guint prop_id, GValue * value, - GParamSpec * pspec) +gst_audio_wsincband_get_property (GObject * object, guint prop_id, + GValue * value, GParamSpec * pspec) { GstAudioWSincBand *self = GST_AUDIO_WSINC_BAND (object); |