From 0016658ace7e5b84095b44ad7bae27218334143b Mon Sep 17 00:00:00 2001 From: Sebastian Dröge Date: Sun, 11 Jan 2009 19:03:38 +0000 Subject: gst/audiofx/: Implement a base class for generic audio FIR filters. Original commit message from CVS: * gst/audiofx/Makefile.am: * gst/audiofx/audiofxbasefirfilter.c: (gst_audio_fx_base_fir_filter_dispose), (gst_audio_fx_base_fir_filter_base_init), (gst_audio_fx_base_fir_filter_class_init), (gst_audio_fx_base_fir_filter_init), (gst_audio_fx_base_fir_filter_push_residue), (gst_audio_fx_base_fir_filter_setup), (gst_audio_fx_base_fir_filter_transform), (gst_audio_fx_base_fir_filter_start), (gst_audio_fx_base_fir_filter_stop), (gst_audio_fx_base_fir_filter_query), (gst_audio_fx_base_fir_filter_query_type), (gst_audio_fx_base_fir_filter_event), (gst_audio_fx_base_fir_filter_set_kernel): * gst/audiofx/audiofxbasefirfilter.h: * gst/audiofx/audiofxbaseiirfilter.c: Implement a base class for generic audio FIR filters. * gst/audiofx/audiowsincband.c: (gst_gst_audio_wsincband_mode_get_type), (gst_gst_audio_wsincband_window_get_type), (gst_audio_wsincband_base_init), (gst_audio_wsincband_class_init), (gst_audio_wsincband_init), (gst_audio_wsincband_build_kernel), (gst_audio_wsincband_setup), (gst_audio_wsincband_set_property), (gst_audio_wsincband_get_property): * gst/audiofx/audiowsincband.h: * gst/audiofx/audiowsinclimit.c: (gst_audio_wsinclimit_mode_get_type), (gst_audio_wsinclimit_window_get_type), (gst_audio_wsinclimit_base_init), (gst_audio_wsinclimit_class_init), (gst_audio_wsinclimit_init), (gst_audio_wsinclimit_build_kernel), (gst_audio_wsinclimit_setup), (gst_audio_wsinclimit_set_property), (gst_audio_wsinclimit_get_property): * gst/audiofx/audiowsinclimit.h: * tests/check/elements/audiowsincband.c: (GST_START_TEST): * tests/check/elements/audiowsinclimit.c: (GST_START_TEST): Use this new base class for audiowsincband and audiowsinclimit. Also cleanup both elements. --- ChangeLog | 43 +++ gst/audiofx/Makefile.am | 2 + gst/audiofx/audiofxbasefirfilter.c | 527 ++++++++++++++++++++++++++++++ gst/audiofx/audiofxbasefirfilter.h | 81 +++++ gst/audiofx/audiofxbaseiirfilter.c | 2 +- gst/audiofx/audiowsincband.c | 570 ++++----------------------------- gst/audiofx/audiowsincband.h | 23 +- gst/audiofx/audiowsinclimit.c | 568 ++++---------------------------- gst/audiofx/audiowsinclimit.h | 25 +- tests/check/elements/audiowsincband.c | 14 + tests/check/elements/audiowsinclimit.c | 10 + 11 files changed, 831 insertions(+), 1034 deletions(-) create mode 100644 gst/audiofx/audiofxbasefirfilter.c create mode 100644 gst/audiofx/audiofxbasefirfilter.h diff --git a/ChangeLog b/ChangeLog index 120c3e3ea..12343bdfd 100644 --- a/ChangeLog +++ b/ChangeLog @@ -1,3 +1,46 @@ +2009-01-11 Sebastian Dröge + + * gst/audiofx/Makefile.am: + * gst/audiofx/audiofxbasefirfilter.c: + (gst_audio_fx_base_fir_filter_dispose), + (gst_audio_fx_base_fir_filter_base_init), + (gst_audio_fx_base_fir_filter_class_init), + (gst_audio_fx_base_fir_filter_init), + (gst_audio_fx_base_fir_filter_push_residue), + (gst_audio_fx_base_fir_filter_setup), + (gst_audio_fx_base_fir_filter_transform), + (gst_audio_fx_base_fir_filter_start), + (gst_audio_fx_base_fir_filter_stop), + (gst_audio_fx_base_fir_filter_query), + (gst_audio_fx_base_fir_filter_query_type), + (gst_audio_fx_base_fir_filter_event), + (gst_audio_fx_base_fir_filter_set_kernel): + * gst/audiofx/audiofxbasefirfilter.h: + * gst/audiofx/audiofxbaseiirfilter.c: + Implement a base class for generic audio FIR filters. + + * gst/audiofx/audiowsincband.c: + (gst_gst_audio_wsincband_mode_get_type), + (gst_gst_audio_wsincband_window_get_type), + (gst_audio_wsincband_base_init), (gst_audio_wsincband_class_init), + (gst_audio_wsincband_init), (gst_audio_wsincband_build_kernel), + (gst_audio_wsincband_setup), (gst_audio_wsincband_set_property), + (gst_audio_wsincband_get_property): + * gst/audiofx/audiowsincband.h: + * gst/audiofx/audiowsinclimit.c: + (gst_audio_wsinclimit_mode_get_type), + (gst_audio_wsinclimit_window_get_type), + (gst_audio_wsinclimit_base_init), + (gst_audio_wsinclimit_class_init), (gst_audio_wsinclimit_init), + (gst_audio_wsinclimit_build_kernel), (gst_audio_wsinclimit_setup), + (gst_audio_wsinclimit_set_property), + (gst_audio_wsinclimit_get_property): + * gst/audiofx/audiowsinclimit.h: + * tests/check/elements/audiowsincband.c: (GST_START_TEST): + * tests/check/elements/audiowsinclimit.c: (GST_START_TEST): + Use this new base class for audiowsincband and audiowsinclimit. + Also cleanup both elements. + 2009-01-08 Michael Smith * gst/qtdemux/qtdemux.c: diff --git a/gst/audiofx/Makefile.am b/gst/audiofx/Makefile.am index ac6439bbb..d93d3e9a9 100644 --- a/gst/audiofx/Makefile.am +++ b/gst/audiofx/Makefile.am @@ -12,6 +12,7 @@ libgstaudiofx_la_SOURCES = audiofx.c\ audiofxbaseiirfilter.c \ audiocheblimit.c \ audiochebband.c \ + audiofxbasefirfilter.c \ audiowsincband.c \ audiowsinclimit.c @@ -38,6 +39,7 @@ noinst_HEADERS = audiopanorama.h \ audiofxbaseiirfilter.h \ audiocheblimit.h \ audiochebband.h \ + audiofxbasefirfilter.h \ audiowsincband.h \ audiowsinclimit.h \ math_compat.h diff --git a/gst/audiofx/audiofxbasefirfilter.c b/gst/audiofx/audiofxbasefirfilter.c new file mode 100644 index 000000000..059c2aa35 --- /dev/null +++ b/gst/audiofx/audiofxbasefirfilter.c @@ -0,0 +1,527 @@ +/* -*- c-basic-offset: 2 -*- + * + * GStreamer + * Copyright (C) 1999-2001 Erik Walthinsen + * 2006 Dreamlab Technologies Ltd. + * 2007-2009 Sebastian Dröge + * + * This library is free software; you can redistribute it and/or + * modify it under the terms of the GNU Library General Public + * License as published by the Free Software Foundation; either + * version 2 of the License, or (at your option) any later version. + * + * This library is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * Library General Public License for more details. + * + * You should have received a copy of the GNU Library General Public + * License along with this library; if not, write to the + * Free Software Foundation, Inc., 59 Temple Place - Suite 330, + * Boston, MA 02111-1307, USA. + * + * + * 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 + */ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include +#include +#include +#include +#include + +#include "audiofxbasefirfilter.h" + +#define GST_CAT_DEFAULT gst_audio_fx_base_fir_filter_debug +GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT); + +#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_fx_base_fir_filter_debug, "audiofxbasefirfilter", 0, \ + "FIR filter base class"); + +GST_BOILERPLATE_FULL (GstAudioFXBaseFIRFilter, gst_audio_fx_base_fir_filter, + GstAudioFilter, GST_TYPE_AUDIO_FILTER, DEBUG_INIT); + +static GstFlowReturn gst_audio_fx_base_fir_filter_transform (GstBaseTransform * + base, GstBuffer * inbuf, GstBuffer * outbuf); +static gboolean gst_audio_fx_base_fir_filter_start (GstBaseTransform * base); +static gboolean gst_audio_fx_base_fir_filter_stop (GstBaseTransform * base); +static gboolean gst_audio_fx_base_fir_filter_event (GstBaseTransform * base, + GstEvent * event); +static gboolean gst_audio_fx_base_fir_filter_setup (GstAudioFilter * base, + GstRingBufferSpec * format); + +static gboolean gst_audio_fx_base_fir_filter_query (GstPad * pad, + GstQuery * query); +static const GstQueryType *gst_audio_fx_base_fir_filter_query_type (GstPad * + pad); + +/* Element class */ + +static void +gst_audio_fx_base_fir_filter_dispose (GObject * object) +{ + GstAudioFXBaseFIRFilter *self = GST_AUDIO_FX_BASE_FIR_FILTER (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_audio_fx_base_fir_filter_base_init (gpointer g_class) +{ + GstCaps *caps; + + 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_audio_fx_base_fir_filter_class_init (GstAudioFXBaseFIRFilterClass * klass) +{ + GObjectClass *gobject_class = (GObjectClass *) klass; + GstBaseTransformClass *trans_class = (GstBaseTransformClass *) klass; + GstAudioFilterClass *filter_class = (GstAudioFilterClass *) klass; + + gobject_class->dispose = gst_audio_fx_base_fir_filter_dispose; + + trans_class->transform = + GST_DEBUG_FUNCPTR (gst_audio_fx_base_fir_filter_transform); + trans_class->start = GST_DEBUG_FUNCPTR (gst_audio_fx_base_fir_filter_start); + trans_class->stop = GST_DEBUG_FUNCPTR (gst_audio_fx_base_fir_filter_stop); + trans_class->event = GST_DEBUG_FUNCPTR (gst_audio_fx_base_fir_filter_event); + filter_class->setup = GST_DEBUG_FUNCPTR (gst_audio_fx_base_fir_filter_setup); +} + +static void +gst_audio_fx_base_fir_filter_init (GstAudioFXBaseFIRFilter * self, + GstAudioFXBaseFIRFilterClass * g_class) +{ + self->kernel = NULL; + self->residue = NULL; + + self->next_ts = GST_CLOCK_TIME_NONE; + self->next_off = GST_BUFFER_OFFSET_NONE; + + gst_pad_set_query_function (GST_BASE_TRANSFORM (self)->srcpad, + gst_audio_fx_base_fir_filter_query); + gst_pad_set_query_type_function (GST_BASE_TRANSFORM (self)->srcpad, + gst_audio_fx_base_fir_filter_query_type); +} + +#define DEFINE_PROCESS_FUNC(width,ctype) \ +static void \ +process_##width (GstAudioFXBaseFIRFilter * 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 + +void +gst_audio_fx_base_fir_filter_push_residue (GstAudioFXBaseFIRFilter * 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; + + if (channels == 0 || rate == 0) { + self->residue_length = 0; + return; + } + + /* 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) { + self->residue_length = 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); + self->residue_length = 0; + 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; + self->next_off = GST_BUFFER_OFFSET_END (outbuf); + } + + 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"); + } + + self->residue_length = 0; +} + +/* GstAudioFilter vmethod implementations */ + +/* get notified of caps and plug in the correct process function */ +static gboolean +gst_audio_fx_base_fir_filter_setup (GstAudioFilter * base, + GstRingBufferSpec * format) +{ + GstAudioFXBaseFIRFilter *self = GST_AUDIO_FX_BASE_FIR_FILTER (base); + gboolean ret = TRUE; + + if (self->residue) { + gst_audio_fx_base_fir_filter_push_residue (self); + g_free (self->residue); + self->residue = NULL; + self->residue_length = 0; + self->next_ts = GST_CLOCK_TIME_NONE; + self->next_off = GST_BUFFER_OFFSET_NONE; + } + + if (format->width == 32) + self->process = (GstAudioFXBaseFIRFilterProcessFunc) process_32; + else if (format->width == 64) + self->process = (GstAudioFXBaseFIRFilterProcessFunc) process_64; + else + ret = FALSE; + + return TRUE; +} + +/* GstBaseTransform vmethod implementations */ + +static GstFlowReturn +gst_audio_fx_base_fir_filter_transform (GstBaseTransform * base, + GstBuffer * inbuf, GstBuffer * outbuf) +{ + GstAudioFXBaseFIRFilter *self = GST_AUDIO_FX_BASE_FIR_FILTER (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 = 0; + + timestamp = GST_BUFFER_TIMESTAMP (outbuf); + if (!GST_CLOCK_TIME_IS_VALID (timestamp)) { + GST_ERROR_OBJECT (self, "Invalid timestamp"); + return GST_FLOW_ERROR; + } + + gst_object_sync_values (G_OBJECT (self), timestamp); + + g_return_val_if_fail (self->kernel != NULL, GST_FLOW_ERROR); + g_return_val_if_fail (channels != 0, GST_FLOW_ERROR); + + if (!self->residue) + self->residue = g_new0 (gdouble, self->kernel_length * channels); + + /* Reset the residue if already existing on discont buffers */ + if (GST_BUFFER_IS_DISCONT (inbuf) || (GST_CLOCK_TIME_IS_VALID (self->next_ts) + && timestamp - gst_util_uint64_scale (MIN (self->latency, + self->residue_length / channels), GST_SECOND, + rate) - self->next_ts > 5 * GST_MSECOND)) { + GST_DEBUG_OBJECT (self, "Discontinuity detected - flushing"); + if (GST_CLOCK_TIME_IS_VALID (self->next_ts)) + gst_audio_fx_base_fir_filter_push_residue (self); + self->residue_length = 0; + self->next_ts = timestamp; + self->next_off = GST_BUFFER_OFFSET (inbuf); + } else if (!GST_CLOCK_TIME_IS_VALID (self->next_ts)) { + self->next_ts = timestamp; + self->next_off = GST_BUFFER_OFFSET (inbuf); + } + + /* Calculate the number of samples we can push out now without outputting + * latency zeros in the beginning */ + diff = self->latency * 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) { + return GST_BASE_TRANSFORM_FLOW_DROPPED; + } + + GST_BUFFER_TIMESTAMP (outbuf) = self->next_ts; + GST_BUFFER_DURATION (outbuf) = + gst_util_uint64_scale (output_samples / channels, GST_SECOND, rate); + GST_BUFFER_OFFSET (outbuf) = self->next_off; + if (GST_BUFFER_OFFSET_IS_VALID (outbuf)) + GST_BUFFER_OFFSET_END (outbuf) = self->next_off + output_samples / channels; + else + GST_BUFFER_OFFSET_END (outbuf) = GST_BUFFER_OFFSET_NONE; + + if (output_samples < input_samples) { + GST_BUFFER_DATA (outbuf) += + diff * (GST_AUDIO_FILTER (self)->format.width / 8); + GST_BUFFER_SIZE (outbuf) -= + diff * (GST_AUDIO_FILTER (self)->format.width / 8); + } + + self->next_ts += GST_BUFFER_DURATION (outbuf); + self->next_off = GST_BUFFER_OFFSET_END (outbuf); + + 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); + + return GST_FLOW_OK; +} + +static gboolean +gst_audio_fx_base_fir_filter_start (GstBaseTransform * base) +{ + GstAudioFXBaseFIRFilter *self = GST_AUDIO_FX_BASE_FIR_FILTER (base); + + self->residue_length = 0; + self->next_ts = GST_CLOCK_TIME_NONE; + self->next_off = GST_BUFFER_OFFSET_NONE; + + return TRUE; +} + +static gboolean +gst_audio_fx_base_fir_filter_stop (GstBaseTransform * base) +{ + GstAudioFXBaseFIRFilter *self = GST_AUDIO_FX_BASE_FIR_FILTER (base); + + g_free (self->residue); + self->residue = NULL; + + return TRUE; +} + +static gboolean +gst_audio_fx_base_fir_filter_query (GstPad * pad, GstQuery * query) +{ + GstAudioFXBaseFIRFilter *self = + GST_AUDIO_FX_BASE_FIR_FILTER (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 (rate == 0) { + res = FALSE; + } else 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 = gst_util_uint64_scale (self->latency, GST_SECOND, rate); + + 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 * +gst_audio_fx_base_fir_filter_query_type (GstPad * pad) +{ + static const GstQueryType types[] = { + GST_QUERY_LATENCY, + 0 + }; + + return types; +} + +static gboolean +gst_audio_fx_base_fir_filter_event (GstBaseTransform * base, GstEvent * event) +{ + GstAudioFXBaseFIRFilter *self = GST_AUDIO_FX_BASE_FIR_FILTER (base); + + switch (GST_EVENT_TYPE (event)) { + case GST_EVENT_EOS: + gst_audio_fx_base_fir_filter_push_residue (self); + self->next_ts = GST_CLOCK_TIME_NONE; + self->next_off = GST_BUFFER_OFFSET_NONE; + break; + default: + break; + } + + return GST_BASE_TRANSFORM_CLASS (parent_class)->event (base, event); +} + +void +gst_audio_fx_base_fir_filter_set_kernel (GstAudioFXBaseFIRFilter * self, + gdouble * kernel, guint kernel_length, guint64 latency) +{ + g_return_if_fail (kernel != NULL); + g_return_if_fail (self != NULL); + + GST_BASE_TRANSFORM_LOCK (self); + if (self->residue) { + gst_audio_fx_base_fir_filter_push_residue (self); + self->next_ts = GST_CLOCK_TIME_NONE; + self->next_off = GST_BUFFER_OFFSET_NONE; + self->residue_length = 0; + } + + g_free (self->kernel); + g_free (self->residue); + + self->kernel = kernel; + self->kernel_length = kernel_length; + + if (GST_AUDIO_FILTER (self)->format.channels) { + self->residue = + g_new0 (gdouble, + kernel_length * GST_AUDIO_FILTER (self)->format.channels); + self->residue_length = 0; + } + + if (self->latency != latency) { + self->latency = latency; + gst_element_post_message (GST_ELEMENT (self), + gst_message_new_latency (GST_OBJECT (self))); + } + + GST_BASE_TRANSFORM_UNLOCK (self); +} diff --git a/gst/audiofx/audiofxbasefirfilter.h b/gst/audiofx/audiofxbasefirfilter.h new file mode 100644 index 000000000..52f424827 --- /dev/null +++ b/gst/audiofx/audiofxbasefirfilter.h @@ -0,0 +1,81 @@ +/* -*- c-basic-offset: 2 -*- + * + * GStreamer + * Copyright (C) 1999-2001 Erik Walthinsen + * 2006 Dreamlab Technologies Ltd. + * 2009 Sebastian Dröge + * + * This library is free software; you can redistribute it and/or + * modify it under the terms of the GNU Library General Public + * License as published by the Free Software Foundation; either + * version 2 of the License, or (at your option) any later version. + * + * This library is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * Library General Public License for more details. + * + * You should have received a copy of the GNU Library General Public + * License along with this library; if not, write to the + * Free Software Foundation, Inc., 59 Temple Place - Suite 330, + * Boston, MA 02111-1307, USA. + * + */ + +#ifndef __GST_AUDIO_FX_BASE_FIR_FILTER_H__ +#define __GST_AUDIO_FX_BASE_FIR_FILTER_H__ + +#include +#include + +G_BEGIN_DECLS + +#define GST_TYPE_AUDIO_FX_BASE_FIR_FILTER \ + (gst_audio_fx_base_fir_filter_get_type()) +#define GST_AUDIO_FX_BASE_FIR_FILTER(obj) \ + (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIO_FX_BASE_FIR_FILTER,GstAudioFXBaseFIRFilter)) +#define GST_AUDIO_FX_BASE_FIR_FILTER_CLASS(klass) \ + (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_AUDIO_FX_BASE_FIR_FILTER,GstAudioFXBaseFIRFilterClass)) +#define GST_IS_AUDIO_FX_BASE_FIR_FILTER(obj) \ + (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIO_FX_BASE_FIR_FILTER)) +#define GST_IS_AUDIO_FX_BASE_FIR_FILTER_CLASS(klass) \ + (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_AUDIO_FX_BASE_FIR_FILTER)) + +typedef struct _GstAudioFXBaseFIRFilter GstAudioFXBaseFIRFilter; +typedef struct _GstAudioFXBaseFIRFilterClass GstAudioFXBaseFIRFilterClass; + +typedef void (*GstAudioFXBaseFIRFilterProcessFunc) (GstAudioFXBaseFIRFilter *, guint8 *, guint8 *, guint); + +/** + * GstAudioFXBaseFIRFilter: + * + * Opaque data structure. + */ +struct _GstAudioFXBaseFIRFilter { + GstAudioFilter element; + + /* < private > */ + GstAudioFXBaseFIRFilterProcessFunc process; + + gdouble *kernel; /* filter kernel */ + guint kernel_length; /* length of the filter kernel */ + gdouble *residue; /* buffer for left-over samples from previous buffer */ + guint residue_length; + + guint64 latency; + + GstClockTime next_ts; + guint64 next_off; +}; + +struct _GstAudioFXBaseFIRFilterClass { + GstAudioFilterClass parent_class; +}; + +GType gst_audio_fx_base_fir_filter_get_type (void); +void gst_audio_fx_base_fir_filter_set_kernel (GstAudioFXBaseFIRFilter *filter, gdouble *kernel, guint kernel_length, guint64 latency); +void gst_audio_fx_base_fir_filter_push_residue (GstAudioFXBaseFIRFilter *filter); + +G_END_DECLS + +#endif /* __GST_AUDIO_FX_BASE_FIR_FILTER_H__ */ diff --git a/gst/audiofx/audiofxbaseiirfilter.c b/gst/audiofx/audiofxbaseiirfilter.c index 29cb2440c..4571a2daf 100644 --- a/gst/audiofx/audiofxbaseiirfilter.c +++ b/gst/audiofx/audiofxbaseiirfilter.c @@ -43,7 +43,7 @@ GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT); " channels = (int) [ 1, MAX ]" #define DEBUG_INIT(bla) \ - GST_DEBUG_CATEGORY_INIT (gst_audio_fx_base_iir_filter_debug, "audiobaseiirfilter", 0, "Audio IIR Filter Base Class"); + GST_DEBUG_CATEGORY_INIT (gst_audio_fx_base_iir_filter_debug, "audiofxbaseiirfilter", 0, "Audio IIR Filter Base Class"); GST_BOILERPLATE_FULL (GstAudioFXBaseIIRFilter, gst_audio_fx_base_iir_filter, GstAudioFilter, GST_TYPE_AUDIO_FILTER, diff --git a/gst/audiofx/audiowsincband.c b/gst/audiofx/audiowsincband.c index 60848b1bc..620343084 100644 --- a/gst/audiofx/audiowsincband.c +++ b/gst/audiofx/audiowsincband.c @@ -3,7 +3,7 @@ * GStreamer * Copyright (C) 1999-2001 Erik Walthinsen * 2006 Dreamlab Technologies Ltd. - * 2007 Sebastian Dröge + * 2007-2009 Sebastian Dröge * * 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 , " - "Steven W. Smith, " - "Dreamlab Technologies Ltd. , " - "Sebastian Dröge "); - -/* 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 , " + "Steven W. Smith, " + "Dreamlab Technologies Ltd. , " + "Sebastian Dröge "); } 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); diff --git a/gst/audiofx/audiowsincband.h b/gst/audiofx/audiowsincband.h index d99c788d7..977a41f04 100644 --- a/gst/audiofx/audiowsincband.h +++ b/gst/audiofx/audiowsincband.h @@ -3,6 +3,7 @@ * GStreamer * Copyright (C) 1999-2001 Erik Walthinsen * 2006 Dreamlab Technologies Ltd. + * 2007-2009 Sebastian Dröge * * This library is free software; you can redistribute it and/or * modify it under the terms of the GNU Library General Public @@ -33,10 +34,12 @@ #include #include +#include "audiofxbasefirfilter.h" + G_BEGIN_DECLS #define GST_TYPE_AUDIO_WSINC_BAND \ - (audio_wsincband_get_type()) + (gst_audio_wsincband_get_type()) #define GST_AUDIO_WSINC_BAND(obj) \ (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIO_WSINC_BAND,GstAudioWSincBand)) #define GST_AUDIO_WSINC_BAND_CLASS(klass) \ @@ -49,38 +52,26 @@ G_BEGIN_DECLS typedef struct _GstAudioWSincBand GstAudioWSincBand; typedef struct _GstAudioWSincBandClass GstAudioWSincBandClass; -typedef void (*GstAudioWSincBandProcessFunc) (GstAudioWSincBand *, guint8 *, guint8 *, guint); - /** * GstAudioWSincBand: * * Opaque data structure. */ struct _GstAudioWSincBand { - GstAudioFilter element; + GstAudioFXBaseFIRFilter parent; /* < private > */ - GstAudioWSincBandProcessFunc 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 _GstAudioWSincBandClass { - GstAudioFilterClass parent_class; + GstAudioFilterClass parent; }; -GType audio_wsincband_get_type (void); +GType gst_audio_wsincband_get_type (void); G_END_DECLS diff --git a/gst/audiofx/audiowsinclimit.c b/gst/audiofx/audiowsinclimit.c index 109b89bde..68e8522c8 100644 --- a/gst/audiofx/audiowsinclimit.c +++ b/gst/audiofx/audiowsinclimit.c @@ -3,7 +3,7 @@ * GStreamer * Copyright (C) 1999-2001 Erik Walthinsen * 2006 Dreamlab Technologies Ltd. - * 2007 Sebastian Dröge + * 2007-2009 Sebastian Dröge * * 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 , " - "Steven W. Smith, " - "Dreamlab Technologies Ltd. , " - "Sebastian Dröge "); - -/* 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 , " + "Steven W. Smith, " + "Dreamlab Technologies Ltd. , " + "Sebastian Dröge "); } 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); diff --git a/gst/audiofx/audiowsinclimit.h b/gst/audiofx/audiowsinclimit.h index b781a422b..d30b39384 100644 --- a/gst/audiofx/audiowsinclimit.h +++ b/gst/audiofx/audiowsinclimit.h @@ -3,6 +3,7 @@ * GStreamer * Copyright (C) 1999-2001 Erik Walthinsen * 2006 Dreamlab Technologies Ltd. + * 2007-2009 Sebastian Dröge * * This library is free software; you can redistribute it and/or * modify it under the terms of the GNU Library General Public @@ -33,10 +34,12 @@ #include #include +#include "audiofxbasefirfilter.h" + G_BEGIN_DECLS #define GST_TYPE_AUDIO_WSINC_LIMIT \ - (audio_wsinclimit_get_type()) + (gst_audio_wsinclimit_get_type()) #define GST_AUDIO_WSINC_LIMIT(obj) \ (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIO_WSINC_LIMIT,GstAudioWSincLimit)) #define GST_AUDIO_WSINC_LIMIT_CLASS(klass) \ @@ -49,38 +52,26 @@ G_BEGIN_DECLS typedef struct _GstAudioWSincLimit GstAudioWSincLimit; typedef struct _GstAudioWSincLimitClass GstAudioWSincLimitClass; -typedef void (*GstAudioWSincLimitProcessFunc) (GstAudioWSincLimit *, guint8 *, guint8 *, guint); - /** * GstAudioWSincLimit: * * Opaque data structure. */ struct _GstAudioWSincLimit { - GstAudioFilter element; + GstAudioFXBaseFIRFilter parent; /* < private > */ - GstAudioWSincLimitProcessFunc 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; + gint kernel_length; }; struct _GstAudioWSincLimitClass { - GstAudioFilterClass parent_class; + GstAudioFXBaseFIRFilterClass parent; }; -GType audio_wsinclimit_get_type (void); +GType gst_audio_wsinclimit_get_type (void); G_END_DECLS diff --git a/tests/check/elements/audiowsincband.c b/tests/check/elements/audiowsincband.c index c3ea37f03..fc4805dfd 100644 --- a/tests/check/elements/audiowsincband.c +++ b/tests/check/elements/audiowsincband.c @@ -119,6 +119,7 @@ GST_START_TEST (test_32_bp_0hz) g_object_set (G_OBJECT (audiowsincband), "upper-frequency", 44100 / 4.0 + 1000, NULL); inbuffer = gst_buffer_new_and_alloc (1024 * sizeof (gfloat)); + GST_BUFFER_TIMESTAMP (inbuffer) = 0; in = (gfloat *) GST_BUFFER_DATA (inbuffer); for (i = 0; i < 1024; i++) in[i] = 1.0; @@ -180,6 +181,7 @@ GST_START_TEST (test_32_bp_11025hz) g_object_set (G_OBJECT (audiowsincband), "upper-frequency", 44100 / 4.0 + 1000, NULL); inbuffer = gst_buffer_new_and_alloc (1024 * sizeof (gfloat)); + GST_BUFFER_TIMESTAMP (inbuffer) = 0; in = (gfloat *) GST_BUFFER_DATA (inbuffer); for (i = 0; i < 1024; i += 4) { in[i] = 0.0; @@ -246,6 +248,7 @@ GST_START_TEST (test_32_bp_22050hz) g_object_set (G_OBJECT (audiowsincband), "upper-frequency", 44100 / 4.0 + 1000, NULL); inbuffer = gst_buffer_new_and_alloc (1024 * sizeof (gfloat)); + GST_BUFFER_TIMESTAMP (inbuffer) = 0; in = (gfloat *) GST_BUFFER_DATA (inbuffer); for (i = 0; i < 1024; i += 2) { in[i] = 1.0; @@ -309,6 +312,7 @@ GST_START_TEST (test_32_br_0hz) g_object_set (G_OBJECT (audiowsincband), "upper-frequency", 44100 / 4.0 + 1000, NULL); inbuffer = gst_buffer_new_and_alloc (1024 * sizeof (gfloat)); + GST_BUFFER_TIMESTAMP (inbuffer) = 0; in = (gfloat *) GST_BUFFER_DATA (inbuffer); for (i = 0; i < 1024; i++) in[i] = 1.0; @@ -370,6 +374,7 @@ GST_START_TEST (test_32_br_11025hz) g_object_set (G_OBJECT (audiowsincband), "upper-frequency", 44100 / 4.0 + 1000, NULL); inbuffer = gst_buffer_new_and_alloc (1024 * sizeof (gfloat)); + GST_BUFFER_TIMESTAMP (inbuffer) = 0; in = (gfloat *) GST_BUFFER_DATA (inbuffer); for (i = 0; i < 1024; i += 4) { @@ -437,6 +442,7 @@ GST_START_TEST (test_32_br_22050hz) g_object_set (G_OBJECT (audiowsincband), "upper-frequency", 44100 / 4.0 + 1000, NULL); inbuffer = gst_buffer_new_and_alloc (1024 * sizeof (gfloat)); + GST_BUFFER_TIMESTAMP (inbuffer) = 0; in = (gfloat *) GST_BUFFER_DATA (inbuffer); for (i = 0; i < 1024; i += 2) { in[i] = 1.0; @@ -498,6 +504,7 @@ GST_START_TEST (test_32_small_buffer) g_object_set (G_OBJECT (audiowsincband), "upper-frequency", 44100 / 4.0 + 44100 / 16.0, NULL); inbuffer = gst_buffer_new_and_alloc (20 * sizeof (gfloat)); + GST_BUFFER_TIMESTAMP (inbuffer) = 0; in = (gfloat *) GST_BUFFER_DATA (inbuffer); for (i = 0; i < 20; i++) in[i] = 1.0; @@ -553,6 +560,7 @@ GST_START_TEST (test_64_bp_0hz) g_object_set (G_OBJECT (audiowsincband), "upper-frequency", 44100 / 4.0 + 1000, NULL); inbuffer = gst_buffer_new_and_alloc (1024 * sizeof (gdouble)); + GST_BUFFER_TIMESTAMP (inbuffer) = 0; in = (gdouble *) GST_BUFFER_DATA (inbuffer); for (i = 0; i < 1024; i++) in[i] = 1.0; @@ -614,6 +622,7 @@ GST_START_TEST (test_64_bp_11025hz) g_object_set (G_OBJECT (audiowsincband), "upper-frequency", 44100 / 4.0 + 1000, NULL); inbuffer = gst_buffer_new_and_alloc (1024 * sizeof (gdouble)); + GST_BUFFER_TIMESTAMP (inbuffer) = 0; in = (gdouble *) GST_BUFFER_DATA (inbuffer); for (i = 0; i < 1024; i += 4) { in[i] = 0.0; @@ -680,6 +689,7 @@ GST_START_TEST (test_64_bp_22050hz) g_object_set (G_OBJECT (audiowsincband), "upper-frequency", 44100 / 4.0 + 1000, NULL); inbuffer = gst_buffer_new_and_alloc (1024 * sizeof (gdouble)); + GST_BUFFER_TIMESTAMP (inbuffer) = 0; in = (gdouble *) GST_BUFFER_DATA (inbuffer); for (i = 0; i < 1024; i += 2) { in[i] = 1.0; @@ -743,6 +753,7 @@ GST_START_TEST (test_64_br_0hz) g_object_set (G_OBJECT (audiowsincband), "upper-frequency", 44100 / 4.0 + 1000, NULL); inbuffer = gst_buffer_new_and_alloc (1024 * sizeof (gdouble)); + GST_BUFFER_TIMESTAMP (inbuffer) = 0; in = (gdouble *) GST_BUFFER_DATA (inbuffer); for (i = 0; i < 1024; i++) in[i] = 1.0; @@ -804,6 +815,7 @@ GST_START_TEST (test_64_br_11025hz) g_object_set (G_OBJECT (audiowsincband), "upper-frequency", 44100 / 4.0 + 1000, NULL); inbuffer = gst_buffer_new_and_alloc (1024 * sizeof (gdouble)); + GST_BUFFER_TIMESTAMP (inbuffer) = 0; in = (gdouble *) GST_BUFFER_DATA (inbuffer); for (i = 0; i < 1024; i += 4) { @@ -871,6 +883,7 @@ GST_START_TEST (test_64_br_22050hz) g_object_set (G_OBJECT (audiowsincband), "upper-frequency", 44100 / 4.0 + 1000, NULL); inbuffer = gst_buffer_new_and_alloc (1024 * sizeof (gdouble)); + GST_BUFFER_TIMESTAMP (inbuffer) = 0; in = (gdouble *) GST_BUFFER_DATA (inbuffer); for (i = 0; i < 1024; i += 2) { in[i] = 1.0; @@ -932,6 +945,7 @@ GST_START_TEST (test_64_small_buffer) g_object_set (G_OBJECT (audiowsincband), "upper-frequency", 44100 / 4.0 + 44100 / 16.0, NULL); inbuffer = gst_buffer_new_and_alloc (20 * sizeof (gdouble)); + GST_BUFFER_TIMESTAMP (inbuffer) = 0; in = (gdouble *) GST_BUFFER_DATA (inbuffer); for (i = 0; i < 20; i++) in[i] = 1.0; diff --git a/tests/check/elements/audiowsinclimit.c b/tests/check/elements/audiowsinclimit.c index da1760498..3a6280d30 100644 --- a/tests/check/elements/audiowsinclimit.c +++ b/tests/check/elements/audiowsinclimit.c @@ -117,6 +117,7 @@ GST_START_TEST (test_32_lp_0hz) /* cutoff = sampling rate / 4, data = 0 */ g_object_set (G_OBJECT (audiowsinclimit), "cutoff", 44100 / 4.0, NULL); inbuffer = gst_buffer_new_and_alloc (128 * sizeof (gfloat)); + GST_BUFFER_TIMESTAMP (inbuffer) = 0; in = (gfloat *) GST_BUFFER_DATA (inbuffer); for (i = 0; i < 128; i++) in[i] = 1.0; @@ -175,6 +176,7 @@ GST_START_TEST (test_32_lp_22050hz) g_object_set (G_OBJECT (audiowsinclimit), "cutoff", 44100 / 4.0, NULL); inbuffer = gst_buffer_new_and_alloc (128 * sizeof (gfloat)); + GST_BUFFER_TIMESTAMP (inbuffer) = 0; in = (gfloat *) GST_BUFFER_DATA (inbuffer); for (i = 0; i < 128; i += 2) { in[i] = 1.0; @@ -235,6 +237,7 @@ GST_START_TEST (test_32_hp_0hz) g_object_set (G_OBJECT (audiowsinclimit), "cutoff", 44100 / 4.0, NULL); inbuffer = gst_buffer_new_and_alloc (128 * sizeof (gfloat)); + GST_BUFFER_TIMESTAMP (inbuffer) = 0; in = (gfloat *) GST_BUFFER_DATA (inbuffer); for (i = 0; i < 128; i++) in[i] = 1.0; @@ -293,6 +296,7 @@ GST_START_TEST (test_32_hp_22050hz) g_object_set (G_OBJECT (audiowsinclimit), "cutoff", 44100 / 4.0, NULL); inbuffer = gst_buffer_new_and_alloc (128 * sizeof (gfloat)); + GST_BUFFER_TIMESTAMP (inbuffer) = 0; in = (gfloat *) GST_BUFFER_DATA (inbuffer); for (i = 0; i < 128; i += 2) { in[i] = 1.0; @@ -352,6 +356,7 @@ GST_START_TEST (test_32_small_buffer) g_object_set (G_OBJECT (audiowsinclimit), "cutoff", 44100 / 4.0, NULL); inbuffer = gst_buffer_new_and_alloc (20 * sizeof (gfloat)); + GST_BUFFER_TIMESTAMP (inbuffer) = 0; in = (gfloat *) GST_BUFFER_DATA (inbuffer); for (i = 0; i < 20; i++) in[i] = 1.0; @@ -398,6 +403,7 @@ GST_START_TEST (test_64_lp_0hz) /* cutoff = sampling rate / 4, data = 0 */ g_object_set (G_OBJECT (audiowsinclimit), "cutoff", 44100 / 4.0, NULL); inbuffer = gst_buffer_new_and_alloc (128 * sizeof (gdouble)); + GST_BUFFER_TIMESTAMP (inbuffer) = 0; in = (gdouble *) GST_BUFFER_DATA (inbuffer); for (i = 0; i < 128; i++) in[i] = 1.0; @@ -456,6 +462,7 @@ GST_START_TEST (test_64_lp_22050hz) g_object_set (G_OBJECT (audiowsinclimit), "cutoff", 44100 / 4.0, NULL); inbuffer = gst_buffer_new_and_alloc (128 * sizeof (gdouble)); + GST_BUFFER_TIMESTAMP (inbuffer) = 0; in = (gdouble *) GST_BUFFER_DATA (inbuffer); for (i = 0; i < 128; i += 2) { in[i] = 1.0; @@ -516,6 +523,7 @@ GST_START_TEST (test_64_hp_0hz) g_object_set (G_OBJECT (audiowsinclimit), "cutoff", 44100 / 4.0, NULL); inbuffer = gst_buffer_new_and_alloc (128 * sizeof (gdouble)); + GST_BUFFER_TIMESTAMP (inbuffer) = 0; in = (gdouble *) GST_BUFFER_DATA (inbuffer); for (i = 0; i < 128; i++) in[i] = 1.0; @@ -574,6 +582,7 @@ GST_START_TEST (test_64_hp_22050hz) g_object_set (G_OBJECT (audiowsinclimit), "cutoff", 44100 / 4.0, NULL); inbuffer = gst_buffer_new_and_alloc (128 * sizeof (gdouble)); + GST_BUFFER_TIMESTAMP (inbuffer) = 0; in = (gdouble *) GST_BUFFER_DATA (inbuffer); for (i = 0; i < 128; i += 2) { in[i] = 1.0; @@ -633,6 +642,7 @@ GST_START_TEST (test_64_small_buffer) g_object_set (G_OBJECT (audiowsinclimit), "cutoff", 44100 / 4.0, NULL); inbuffer = gst_buffer_new_and_alloc (20 * sizeof (gdouble)); + GST_BUFFER_TIMESTAMP (inbuffer) = 0; in = (gdouble *) GST_BUFFER_DATA (inbuffer); for (i = 0; i < 20; i++) in[i] = 1.0; -- cgit v1.2.3