diff options
Diffstat (limited to 'gst/audiofx/audiowsinclimit.c')
-rw-r--r-- | gst/audiofx/audiowsinclimit.c | 568 |
1 files changed, 68 insertions, 500 deletions
diff --git a/gst/audiofx/audiowsinclimit.c b/gst/audiofx/audiowsinclimit.c index 109b89bd..68e8522c 100644 --- a/gst/audiofx/audiowsinclimit.c +++ b/gst/audiofx/audiowsinclimit.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 @@ -72,25 +72,9 @@ #include "audiowsinclimit.h" -#define GST_CAT_DEFAULT audio_wsinclimit_debug +#define GST_CAT_DEFAULT gst_audio_wsinclimit_debug GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT); -static const GstElementDetails audio_wsinclimit_details = -GST_ELEMENT_DETAILS ("Low pass & high pass filter", - "Filter/Effect/Audio", - "Low pass and high pass windowed sinc filter", - "Thomas Vander Stichele <thomas at apestaart dot org>, " - "Steven W. Smith, " - "Dreamlab Technologies Ltd. <mathis.hofer@dreamlab.net>, " - "Sebastian Dröge <slomo@circular-chaos.org>"); - -/* Filter signals and args */ -enum -{ - /* FILL ME */ - LAST_SIGNAL -}; - enum { PROP_0, @@ -106,9 +90,9 @@ enum MODE_HIGH_PASS }; -#define GST_TYPE_AUDIO_WSINC_LIMIT_MODE (audio_wsinclimit_mode_get_type ()) +#define GST_TYPE_AUDIO_WSINC_LIMIT_MODE (gst_audio_wsinclimit_mode_get_type ()) static GType -audio_wsinclimit_mode_get_type (void) +gst_audio_wsinclimit_mode_get_type (void) { static GType gtype = 0; @@ -132,9 +116,9 @@ enum WINDOW_BLACKMAN }; -#define GST_TYPE_AUDIO_WSINC_LIMIT_WINDOW (audio_wsinclimit_window_get_type ()) +#define GST_TYPE_AUDIO_WSINC_LIMIT_WINDOW (gst_audio_wsinclimit_window_get_type ()) static GType -audio_wsinclimit_window_get_type (void) +gst_audio_wsinclimit_window_get_type (void) { static GType gtype = 0; @@ -152,189 +136,91 @@ audio_wsinclimit_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 (audio_wsinclimit_debug, "audiowsinclimit", 0, \ + GST_DEBUG_CATEGORY_INIT (gst_audio_wsinclimit_debug, "audiowsinclimit", 0, \ "Low-pass and High-pass Windowed sinc filter plugin"); -GST_BOILERPLATE_FULL (GstAudioWSincLimit, audio_wsinclimit, GstAudioFilter, - GST_TYPE_AUDIO_FILTER, DEBUG_INIT); +GST_BOILERPLATE_FULL (GstAudioWSincLimit, gst_audio_wsinclimit, GstAudioFilter, + GST_TYPE_AUDIO_FX_BASE_FIR_FILTER, DEBUG_INIT); -static void audio_wsinclimit_set_property (GObject * object, guint prop_id, +static void gst_audio_wsinclimit_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec); -static void audio_wsinclimit_get_property (GObject * object, guint prop_id, +static void gst_audio_wsinclimit_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec); -static GstFlowReturn audio_wsinclimit_transform (GstBaseTransform * base, - GstBuffer * inbuf, GstBuffer * outbuf); -static gboolean audio_wsinclimit_start (GstBaseTransform * base); -static gboolean audio_wsinclimit_event (GstBaseTransform * base, - GstEvent * event); -static gboolean audio_wsinclimit_setup (GstAudioFilter * base, +static gboolean gst_audio_wsinclimit_setup (GstAudioFilter * base, GstRingBufferSpec * format); -static gboolean audio_wsinclimit_query (GstPad * pad, GstQuery * query); -static const GstQueryType *audio_wsinclimit_query_type (GstPad * pad); - /* Element class */ static void -audio_wsinclimit_dispose (GObject * object) -{ - GstAudioWSincLimit *self = GST_AUDIO_WSINC_LIMIT (object); - - if (self->residue) { - g_free (self->residue); - self->residue = NULL; - } - - if (self->kernel) { - g_free (self->kernel); - self->kernel = NULL; - } - - G_OBJECT_CLASS (parent_class)->dispose (object); -} - -static void -audio_wsinclimit_base_init (gpointer g_class) +gst_audio_wsinclimit_base_init (gpointer g_class) { GstElementClass *element_class = GST_ELEMENT_CLASS (g_class); - GstCaps *caps; - - gst_element_class_set_details (element_class, &audio_wsinclimit_details); - caps = gst_caps_from_string (ALLOWED_CAPS); - gst_audio_filter_class_add_pad_templates (GST_AUDIO_FILTER_CLASS (g_class), - caps); - gst_caps_unref (caps); + gst_element_class_set_details_simple (element_class, + "Low pass & high pass filter", "Filter/Effect/Audio", + "Low pass and high pass windowed sinc filter", + "Thomas Vander Stichele <thomas at apestaart dot org>, " + "Steven W. Smith, " + "Dreamlab Technologies Ltd. <mathis.hofer@dreamlab.net>, " + "Sebastian Dröge <sebastian.droege@collabora.co.uk>"); } static void -audio_wsinclimit_class_init (GstAudioWSincLimitClass * klass) +gst_audio_wsinclimit_class_init (GstAudioWSincLimitClass * klass) { - GObjectClass *gobject_class; - GstBaseTransformClass *trans_class; - GstAudioFilterClass *filter_class; - - gobject_class = (GObjectClass *) klass; - trans_class = (GstBaseTransformClass *) klass; - filter_class = (GstAudioFilterClass *) klass; - - gobject_class->set_property = audio_wsinclimit_set_property; - gobject_class->get_property = audio_wsinclimit_get_property; - gobject_class->dispose = audio_wsinclimit_dispose; + GObjectClass *gobject_class = (GObjectClass *) klass; + GstAudioFilterClass *filter_class = (GstAudioFilterClass *) klass; + gobject_class->set_property = gst_audio_wsinclimit_set_property; + gobject_class->get_property = gst_audio_wsinclimit_get_property; /* FIXME: Don't use the complete possible range but restrict the upper boundary * so automatically generated UIs can use a slider */ g_object_class_install_property (gobject_class, PROP_FREQUENCY, g_param_spec_float ("cutoff", "Cutoff", "Cut-off Frequency (Hz)", 0.0, 100000.0, 0.0, - G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE)); + G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (gobject_class, PROP_LENGTH, g_param_spec_int ("length", "Length", "Filter kernel length, will be rounded to the next odd number", - 3, 50000, 101, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE)); + 3, 50000, 101, + G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (gobject_class, PROP_MODE, g_param_spec_enum ("mode", "Mode", "Low pass or high pass mode", GST_TYPE_AUDIO_WSINC_LIMIT_MODE, - MODE_LOW_PASS, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE)); + MODE_LOW_PASS, + G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (gobject_class, PROP_WINDOW, g_param_spec_enum ("window", "Window", "Window function to use", GST_TYPE_AUDIO_WSINC_LIMIT_WINDOW, - WINDOW_HAMMING, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE)); + WINDOW_HAMMING, + G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE | G_PARAM_STATIC_STRINGS)); - trans_class->transform = GST_DEBUG_FUNCPTR (audio_wsinclimit_transform); - trans_class->start = GST_DEBUG_FUNCPTR (audio_wsinclimit_start); - trans_class->event = GST_DEBUG_FUNCPTR (audio_wsinclimit_event); - filter_class->setup = GST_DEBUG_FUNCPTR (audio_wsinclimit_setup); + filter_class->setup = GST_DEBUG_FUNCPTR (gst_audio_wsinclimit_setup); } static void -audio_wsinclimit_init (GstAudioWSincLimit * self, +gst_audio_wsinclimit_init (GstAudioWSincLimit * self, GstAudioWSincLimitClass * g_class) { self->mode = MODE_LOW_PASS; self->window = WINDOW_HAMMING; self->kernel_length = 101; - self->latency = 50; self->cutoff = 0.0; - self->kernel = NULL; - self->residue = NULL; - - self->have_kernel = FALSE; - self->residue_length = 0; - self->next_ts = GST_CLOCK_TIME_NONE; - self->next_off = GST_BUFFER_OFFSET_NONE; - - gst_pad_set_query_function (GST_BASE_TRANSFORM (self)->srcpad, - audio_wsinclimit_query); - gst_pad_set_query_type_function (GST_BASE_TRANSFORM (self)->srcpad, - audio_wsinclimit_query_type); -} - -#define DEFINE_PROCESS_FUNC(width,ctype) \ -static void \ -process_##width (GstAudioWSincLimit * self, g##ctype * src, g##ctype * dst, guint input_samples) \ -{ \ - gint kernel_length = self->kernel_length; \ - gint i, j, k, l; \ - gint channels = GST_AUDIO_FILTER (self)->format.channels; \ - gint res_start; \ - \ - /* convolution */ \ - for (i = 0; i < input_samples; i++) { \ - dst[i] = 0.0; \ - k = i % channels; \ - l = i / channels; \ - for (j = 0; j < kernel_length; j++) \ - if (l < j) \ - dst[i] += \ - self->residue[(kernel_length + l - j) * channels + \ - k] * self->kernel[j]; \ - else \ - dst[i] += src[(l - j) * channels + k] * self->kernel[j]; \ - } \ - \ - /* copy the tail of the current input buffer to the residue, while \ - * keeping parts of the residue if the input buffer is smaller than \ - * the kernel length */ \ - if (input_samples < kernel_length * channels) \ - res_start = kernel_length * channels - input_samples; \ - else \ - res_start = 0; \ - \ - for (i = 0; i < res_start; i++) \ - self->residue[i] = self->residue[i + input_samples]; \ - for (i = res_start; i < kernel_length * channels; i++) \ - self->residue[i] = src[input_samples - kernel_length * channels + i]; \ - \ - self->residue_length += kernel_length * channels - res_start; \ - if (self->residue_length > kernel_length * channels) \ - self->residue_length = kernel_length * channels; \ } -DEFINE_PROCESS_FUNC (32, float); -DEFINE_PROCESS_FUNC (64, double); - -#undef DEFINE_PROCESS_FUNC - static void -audio_wsinclimit_build_kernel (GstAudioWSincLimit * self) +gst_audio_wsinclimit_build_kernel (GstAudioWSincLimit * self) { gint i = 0; gdouble sum = 0.0; gint len = 0; gdouble w; + gdouble *kernel = NULL; len = self->kernel_length; @@ -352,7 +238,7 @@ audio_wsinclimit_build_kernel (GstAudioWSincLimit * self) self->cutoff = CLAMP (self->cutoff, 0.0, GST_AUDIO_FILTER (self)->format.rate / 2); - GST_DEBUG ("audio_wsinclimit_: initializing filter kernel of length %d " + GST_DEBUG ("gst_audio_wsinclimit_: initializing filter kernel of length %d " "with cutoff %.2lf Hz " "for mode %s", len, self->cutoff, @@ -361,365 +247,53 @@ audio_wsinclimit_build_kernel (GstAudioWSincLimit * self) /* fill the kernel */ w = 2 * M_PI * (self->cutoff / GST_AUDIO_FILTER (self)->format.rate); - if (self->kernel) - g_free (self->kernel); - self->kernel = g_new (gdouble, len); + kernel = g_new (gdouble, len); for (i = 0; i < len; ++i) { if (i == len / 2) - self->kernel[i] = w; + kernel[i] = w; else - self->kernel[i] = sin (w * (i - len / 2)) / (i - len / 2); + kernel[i] = sin (w * (i - len / 2)) / (i - len / 2); /* windowing */ if (self->window == WINDOW_HAMMING) - self->kernel[i] *= (0.54 - 0.46 * cos (2 * M_PI * i / len)); + kernel[i] *= (0.54 - 0.46 * cos (2 * M_PI * i / len)); else - self->kernel[i] *= - (0.42 - 0.5 * cos (2 * M_PI * i / len) + + kernel[i] *= (0.42 - 0.5 * cos (2 * M_PI * i / len) + 0.08 * cos (4 * M_PI * i / len)); } /* normalize for unity gain at DC */ for (i = 0; i < len; ++i) - sum += self->kernel[i]; + sum += kernel[i]; for (i = 0; i < len; ++i) - self->kernel[i] /= sum; + kernel[i] /= sum; /* convert to highpass if specified */ if (self->mode == MODE_HIGH_PASS) { for (i = 0; i < len; ++i) - self->kernel[i] = -self->kernel[i]; - self->kernel[len / 2] += 1.0; - } - - /* set up the residue memory space */ - if (!self->residue) { - self->residue = - g_new0 (gdouble, len * GST_AUDIO_FILTER (self)->format.channels); - self->residue_length = 0; - } - - self->have_kernel = TRUE; -} - -static void -audio_wsinclimit_push_residue (GstAudioWSincLimit * self) -{ - GstBuffer *outbuf; - GstFlowReturn res; - gint rate = GST_AUDIO_FILTER (self)->format.rate; - gint channels = GST_AUDIO_FILTER (self)->format.channels; - gint outsize, outsamples; - gint diffsize, diffsamples; - guint8 *in, *out; - - /* Calculate the number of samples and their memory size that - * should be pushed from the residue */ - outsamples = MIN (self->latency, self->residue_length / channels); - outsize = outsamples * channels * (GST_AUDIO_FILTER (self)->format.width / 8); - if (outsize == 0) - return; - - /* Process the difference between latency and residue_length samples - * to start at the actual data instead of starting at the zeros before - * when we only got one buffer smaller than latency */ - diffsamples = self->latency - self->residue_length / channels; - diffsize = - diffsamples * channels * (GST_AUDIO_FILTER (self)->format.width / 8); - if (diffsize > 0) { - in = g_new0 (guint8, diffsize); - out = g_new0 (guint8, diffsize); - self->process (self, in, out, diffsamples * channels); - g_free (in); - g_free (out); - } - - res = gst_pad_alloc_buffer (GST_BASE_TRANSFORM (self)->srcpad, - GST_BUFFER_OFFSET_NONE, outsize, - GST_PAD_CAPS (GST_BASE_TRANSFORM (self)->srcpad), &outbuf); - - if (G_UNLIKELY (res != GST_FLOW_OK)) { - GST_WARNING_OBJECT (self, "failed allocating buffer of %d bytes", outsize); - return; - } - - /* Convolve the residue with zeros to get the actual remaining data */ - in = g_new0 (guint8, outsize); - self->process (self, in, GST_BUFFER_DATA (outbuf), outsamples * channels); - g_free (in); - - /* Set timestamp, offset, etc from the values we - * saved when processing the regular buffers */ - if (GST_CLOCK_TIME_IS_VALID (self->next_ts)) - GST_BUFFER_TIMESTAMP (outbuf) = self->next_ts; - else - GST_BUFFER_TIMESTAMP (outbuf) = 0; - GST_BUFFER_DURATION (outbuf) = - gst_util_uint64_scale (outsamples, GST_SECOND, rate); - self->next_ts += gst_util_uint64_scale (outsamples, GST_SECOND, rate); - - if (self->next_off != GST_BUFFER_OFFSET_NONE) { - GST_BUFFER_OFFSET (outbuf) = self->next_off; - GST_BUFFER_OFFSET_END (outbuf) = self->next_off + outsamples; - } - - GST_DEBUG_OBJECT (self, "Pushing residue buffer of size %d with timestamp: %" - GST_TIME_FORMAT ", duration: %" GST_TIME_FORMAT ", offset: %lld," - " offset_end: %lld, nsamples: %d", GST_BUFFER_SIZE (outbuf), - GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf)), - GST_TIME_ARGS (GST_BUFFER_DURATION (outbuf)), GST_BUFFER_OFFSET (outbuf), - GST_BUFFER_OFFSET_END (outbuf), outsamples); - - res = gst_pad_push (GST_BASE_TRANSFORM (self)->srcpad, outbuf); - - if (G_UNLIKELY (res != GST_FLOW_OK)) { - GST_WARNING_OBJECT (self, "failed to push residue"); + kernel[i] = -kernel[i]; + kernel[len / 2] += 1.0; } + gst_audio_fx_base_fir_filter_set_kernel (GST_AUDIO_FX_BASE_FIR_FILTER (self), + kernel, self->kernel_length, (len - 1) / 2); } /* GstAudioFilter vmethod implementations */ /* get notified of caps and plug in the correct process function */ static gboolean -audio_wsinclimit_setup (GstAudioFilter * base, GstRingBufferSpec * format) +gst_audio_wsinclimit_setup (GstAudioFilter * base, GstRingBufferSpec * format) { GstAudioWSincLimit *self = GST_AUDIO_WSINC_LIMIT (base); - gboolean ret = TRUE; - - if (format->width == 32) - self->process = (GstAudioWSincLimitProcessFunc) process_32; - else if (format->width == 64) - self->process = (GstAudioWSincLimitProcessFunc) process_64; - else - ret = FALSE; - - self->have_kernel = FALSE; - - return TRUE; -} - -/* GstBaseTransform vmethod implementations */ - -static GstFlowReturn -audio_wsinclimit_transform (GstBaseTransform * base, GstBuffer * inbuf, - GstBuffer * outbuf) -{ - GstAudioWSincLimit *self = GST_AUDIO_WSINC_LIMIT (base); - GstClockTime timestamp; - gint channels = GST_AUDIO_FILTER (self)->format.channels; - gint rate = GST_AUDIO_FILTER (self)->format.rate; - gint input_samples = - GST_BUFFER_SIZE (outbuf) / (GST_AUDIO_FILTER (self)->format.width / 8); - gint output_samples = input_samples; - gint diff; - - /* FIXME: subdivide GST_BUFFER_SIZE into small chunks for smooth fades */ - timestamp = GST_BUFFER_TIMESTAMP (outbuf); - if (GST_CLOCK_TIME_IS_VALID (timestamp)) - gst_object_sync_values (G_OBJECT (self), timestamp); - - if (!self->have_kernel) - audio_wsinclimit_build_kernel (self); - - /* Reset the residue if already existing on discont buffers */ - if (GST_BUFFER_IS_DISCONT (inbuf)) { - if (channels && self->residue) - memset (self->residue, 0, channels * - self->kernel_length * sizeof (gdouble)); - self->residue_length = 0; - self->next_ts = GST_CLOCK_TIME_NONE; - self->next_off = GST_BUFFER_OFFSET_NONE; - } - - /* Calculate the number of samples we can push out now without outputting - * kernel_length/2 zeros in the beginning */ - diff = (self->kernel_length / 2) * channels - self->residue_length; - if (diff > 0) - output_samples -= diff; - - self->process (self, GST_BUFFER_DATA (inbuf), GST_BUFFER_DATA (outbuf), - input_samples); - - if (output_samples <= 0) { - /* Drop buffer and save original timestamp/offset for later use */ - if (!GST_CLOCK_TIME_IS_VALID (self->next_ts) - && GST_BUFFER_TIMESTAMP_IS_VALID (outbuf)) - self->next_ts = GST_BUFFER_TIMESTAMP (outbuf); - if (self->next_off == GST_BUFFER_OFFSET_NONE - && GST_BUFFER_OFFSET_IS_VALID (outbuf)) - self->next_off = GST_BUFFER_OFFSET (outbuf); - return GST_BASE_TRANSFORM_FLOW_DROPPED; - } else if (output_samples < input_samples) { - /* First (probably partial) buffer after starting from - * a clean residue. Use stored timestamp/offset here */ - if (GST_CLOCK_TIME_IS_VALID (self->next_ts)) - GST_BUFFER_TIMESTAMP (outbuf) = self->next_ts; - - if (self->next_off != GST_BUFFER_OFFSET_NONE) { - GST_BUFFER_OFFSET (outbuf) = self->next_off; - if (GST_BUFFER_OFFSET_END_IS_VALID (outbuf)) - GST_BUFFER_OFFSET_END (outbuf) = - self->next_off + output_samples / channels; - } else { - /* We dropped no buffer, offset is valid, offset_end must be adjusted by diff */ - if (GST_BUFFER_OFFSET_END_IS_VALID (outbuf)) - GST_BUFFER_OFFSET_END (outbuf) -= diff / channels; - } - - if (GST_BUFFER_DURATION_IS_VALID (outbuf)) - GST_BUFFER_DURATION (outbuf) -= - gst_util_uint64_scale (diff, GST_SECOND, channels * rate); - - GST_BUFFER_DATA (outbuf) += - diff * (GST_AUDIO_FILTER (self)->format.width / 8); - GST_BUFFER_SIZE (outbuf) -= - diff * (GST_AUDIO_FILTER (self)->format.width / 8); - } else { - GstClockTime ts_latency = - gst_util_uint64_scale (self->latency, GST_SECOND, rate); - - /* Normal buffer, adjust timestamp/offset/etc by latency */ - if (GST_BUFFER_TIMESTAMP (outbuf) < ts_latency) { - GST_WARNING_OBJECT (self, "GST_BUFFER_TIMESTAMP (outbuf) < latency"); - GST_BUFFER_TIMESTAMP (outbuf) = 0; - } else { - GST_BUFFER_TIMESTAMP (outbuf) -= ts_latency; - } - - if (GST_BUFFER_OFFSET_IS_VALID (outbuf)) { - if (GST_BUFFER_OFFSET (outbuf) > self->latency) { - GST_BUFFER_OFFSET (outbuf) -= self->latency; - } else { - GST_WARNING_OBJECT (self, "GST_BUFFER_OFFSET (outbuf) < latency"); - GST_BUFFER_OFFSET (outbuf) = 0; - } - } - - if (GST_BUFFER_OFFSET_END_IS_VALID (outbuf)) { - if (GST_BUFFER_OFFSET_END (outbuf) > self->latency) { - GST_BUFFER_OFFSET_END (outbuf) -= self->latency; - } else { - GST_WARNING_OBJECT (self, "GST_BUFFER_OFFSET_END (outbuf) < latency"); - GST_BUFFER_OFFSET_END (outbuf) = 0; - } - } - } - - GST_DEBUG_OBJECT (self, "Pushing buffer of size %d with timestamp: %" - GST_TIME_FORMAT ", duration: %" GST_TIME_FORMAT ", offset: %lld," - " offset_end: %lld, nsamples: %d", GST_BUFFER_SIZE (outbuf), - GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf)), - GST_TIME_ARGS (GST_BUFFER_DURATION (outbuf)), GST_BUFFER_OFFSET (outbuf), - GST_BUFFER_OFFSET_END (outbuf), output_samples / channels); - - self->next_ts = GST_BUFFER_TIMESTAMP (outbuf) + GST_BUFFER_DURATION (outbuf); - self->next_off = GST_BUFFER_OFFSET_END (outbuf); - - return GST_FLOW_OK; -} - -static gboolean -audio_wsinclimit_start (GstBaseTransform * base) -{ - GstAudioWSincLimit *self = GST_AUDIO_WSINC_LIMIT (base); - gint channels = GST_AUDIO_FILTER (self)->format.channels; - - /* Reset the residue if already existing */ - if (channels && self->residue) - memset (self->residue, 0, channels * - self->kernel_length * sizeof (gdouble)); - - self->residue_length = 0; - self->next_ts = GST_CLOCK_TIME_NONE; - self->next_off = GST_BUFFER_OFFSET_NONE; - - return TRUE; -} - -static gboolean -audio_wsinclimit_query (GstPad * pad, GstQuery * query) -{ - GstAudioWSincLimit *self = GST_AUDIO_WSINC_LIMIT (gst_pad_get_parent (pad)); - gboolean res = TRUE; - - switch (GST_QUERY_TYPE (query)) { - case GST_QUERY_LATENCY: - { - GstClockTime min, max; - gboolean live; - guint64 latency; - GstPad *peer; - gint rate = GST_AUDIO_FILTER (self)->format.rate; - - if ((peer = gst_pad_get_peer (GST_BASE_TRANSFORM (self)->sinkpad))) { - if ((res = gst_pad_query (peer, query))) { - gst_query_parse_latency (query, &live, &min, &max); - - GST_DEBUG_OBJECT (self, "Peer latency: min %" - GST_TIME_FORMAT " max %" GST_TIME_FORMAT, - GST_TIME_ARGS (min), GST_TIME_ARGS (max)); - - /* add our own latency */ - latency = - (rate != 0) ? gst_util_uint64_scale (self->latency, GST_SECOND, - rate) : 0; - - GST_DEBUG_OBJECT (self, "Our latency: %" - GST_TIME_FORMAT, GST_TIME_ARGS (latency)); - - min += latency; - if (max != GST_CLOCK_TIME_NONE) - max += latency; - - GST_DEBUG_OBJECT (self, "Calculated total latency : min %" - GST_TIME_FORMAT " max %" GST_TIME_FORMAT, - GST_TIME_ARGS (min), GST_TIME_ARGS (max)); - - gst_query_set_latency (query, live, min, max); - } - gst_object_unref (peer); - } - break; - } - default: - res = gst_pad_query_default (pad, query); - break; - } - gst_object_unref (self); - return res; -} - -static const GstQueryType * -audio_wsinclimit_query_type (GstPad * pad) -{ - static const GstQueryType types[] = { - GST_QUERY_LATENCY, - 0 - }; - - return types; -} - -static gboolean -audio_wsinclimit_event (GstBaseTransform * base, GstEvent * event) -{ - GstAudioWSincLimit *self = GST_AUDIO_WSINC_LIMIT (base); - - switch (GST_EVENT_TYPE (event)) { - case GST_EVENT_EOS: - audio_wsinclimit_push_residue (self); - break; - default: - break; - } + gst_audio_wsinclimit_build_kernel (self); - return GST_BASE_TRANSFORM_CLASS (parent_class)->event (base, event); + return GST_AUDIO_FILTER_CLASS (parent_class)->setup (base, format); } static void -audio_wsinclimit_set_property (GObject * object, guint prop_id, +gst_audio_wsinclimit_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec) { GstAudioWSincLimit *self = GST_AUDIO_WSINC_LIMIT (object); @@ -730,43 +304,37 @@ audio_wsinclimit_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_wsinclimit_push_residue (self); - g_free (self->residue); - self->residue = NULL; - } + gst_audio_fx_base_fir_filter_push_residue (GST_AUDIO_FX_BASE_FIR_FILTER + (self)); self->kernel_length = val; - self->latency = val / 2; - audio_wsinclimit_build_kernel (self); - gst_element_post_message (GST_ELEMENT (self), - gst_message_new_latency (GST_OBJECT (self))); + gst_audio_wsinclimit_build_kernel (self); } - GST_BASE_TRANSFORM_UNLOCK (self); + GST_OBJECT_UNLOCK (self); break; } case PROP_FREQUENCY: - GST_BASE_TRANSFORM_LOCK (self); + GST_OBJECT_LOCK (self); self->cutoff = g_value_get_float (value); - audio_wsinclimit_build_kernel (self); - GST_BASE_TRANSFORM_UNLOCK (self); + gst_audio_wsinclimit_build_kernel (self); + GST_OBJECT_UNLOCK (self); break; case PROP_MODE: - GST_BASE_TRANSFORM_LOCK (self); + GST_OBJECT_LOCK (self); self->mode = g_value_get_enum (value); - audio_wsinclimit_build_kernel (self); - GST_BASE_TRANSFORM_UNLOCK (self); + gst_audio_wsinclimit_build_kernel (self); + GST_OBJECT_UNLOCK (self); break; case PROP_WINDOW: - GST_BASE_TRANSFORM_LOCK (self); + GST_OBJECT_LOCK (self); self->window = g_value_get_enum (value); - audio_wsinclimit_build_kernel (self); - GST_BASE_TRANSFORM_UNLOCK (self); + gst_audio_wsinclimit_build_kernel (self); + GST_OBJECT_UNLOCK (self); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); @@ -775,8 +343,8 @@ audio_wsinclimit_set_property (GObject * object, guint prop_id, } static void -audio_wsinclimit_get_property (GObject * object, guint prop_id, GValue * value, - GParamSpec * pspec) +gst_audio_wsinclimit_get_property (GObject * object, guint prop_id, + GValue * value, GParamSpec * pspec) { GstAudioWSincLimit *self = GST_AUDIO_WSINC_LIMIT (object); |