slomo gst-plugins-good: gst-plugins-good/ gst-plugins-good/gst/audiofx/ gst-plugins-good/tests/check/elements/
[email protected] Sun, 11 Jan 2009 11:03:53 -0800 (PST)
| Newsgroups | gmane.comp.video.gstreamer.cvs |
|---|---|
| Message-ID | <[email protected]> |
--===============4753564101550686637==
CVS Root: /cvs/gstreamer
Module: gst-plugins-good
Changes by: slomo
Date: Sun Jan 11 2009 19:03:53 UTC
Log message:
* 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.
Modified files:
. : ChangeLog
gst/audiofx : Makefile.am audiofxbaseiirfilter.c
audiowsincband.c audiowsincband.h
audiowsinclimit.c audiowsinclimit.h
tests/check/elements: audiowsincband.c audiowsinclimit.c
Added files:
gst/audiofx : audiofxbasefirfilter.c audiofxbasefirfilter.h
Links:
http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gst-plugins-good/ChangeLog.diff?r1=1.3890&r2=1.3891
http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gst-plugins-good/gst/audiofx/Makefile.am.diff?r1=1.16&r2=1.17
http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gst-plugins-good/gst/audiofx/audiofxbasefirfilter.c?rev=1.1&content-type=text/vnd.viewcvs-markup
http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gst-plugins-good/gst/audiofx/audiofxbasefirfilter.h?rev=1.1&content-type=text/vnd.viewcvs-markup
http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gst-plugins-good/gst/audiofx/audiofxbaseiirfilter.c.diff?r1=1.1&r2=1.2
http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gst-plugins-good/gst/audiofx/audiowsincband.c.diff?r1=1.40&r2=1.41
http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gst-plugins-good/gst/audiofx/audiowsincband.h.diff?r1=1.8&r2=1.9
http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gst-plugins-good/gst/audiofx/audiowsinclimit.c.diff?r1=1.46&r2=1.47
http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gst-plugins-good/gst/audiofx/audiowsinclimit.h.diff?r1=1.10&r2=1.11
http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gst-plugins-good/tests/check/elements/audiowsincband.c.diff?r1=1.5&r2=1.6
http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gst-plugins-good/tests/check/elements/audiowsinclimit.c.diff?r1=1.5&r2=1.6
====Begin Diffs====
Index: ChangeLog
===================================================================
RCS file: /cvs/gstreamer/gst-plugins-good/ChangeLog,v
retrieving revision 1.3890
retrieving revision 1.3891
diff -u -d -r1.3890 -r1.3891
--- ChangeLog 8 Jan 2009 18:17:12 -0000 1.3890
+++ ChangeLog 11 Jan 2009 19:03:36 -0000 1.3891
@@ -1,3 +1,46 @@
+2009-01-11 Sebastian Dröge <[email protected]>
+
+ * 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 <[email protected]>
* gst/qtdemux/qtdemux.c:
Index: Makefile.am
RCS file: /cvs/gstreamer/gst-plugins-good/gst/audiofx/Makefile.am,v
retrieving revision 1.16
retrieving revision 1.17
diff -u -d -r1.16 -r1.17
--- Makefile.am 5 Jan 2009 10:13:28 -0000 1.16
+++ Makefile.am 11 Jan 2009 19:03:37 -0000 1.17
@@ -12,6 +12,7 @@
audiofxbaseiirfilter.c \
audiocheblimit.c \
audiochebband.c \
+ audiofxbasefirfilter.c \
audiowsincband.c \
audiowsinclimit.c
@@ -38,6 +39,7 @@
audiofxbaseiirfilter.h \
audiocheblimit.h \
audiochebband.h \
+ audiofxbasefirfilter.h \
audiowsincband.h \
audiowsinclimit.h \
math_compat.h
--- NEW FILE: audiofxbasefirfilter.c ---
/* -*- c-basic-offset: 2 -*-
*
* GStreamer
* Copyright (C) 1999-2001 Erik Walthinsen <[email protected]>
* 2006 Dreamlab Technologies Ltd. <[email protected]>
* 2007-2009 Sebastian Dröge <[email protected]>
*
* 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 <string.h>
#include <math.h>
#include <gst/gst.h>
#include <gst/audio/gstaudiofilter.h>
#include <gst/controller/gstcontroller.h>
#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);
}
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);
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);
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) {
/* 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);
/* 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);
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;
gst_audio_fx_base_fir_filter_push_residue (self);
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;
ret = FALSE;
return TRUE;
/* GstBaseTransform vmethod implementations */
static GstFlowReturn
gst_audio_fx_base_fir_filter_transform (GstBaseTransform * base,
GstBuffer * inbuf, GstBuffer * outbuf)
GstClockTime timestamp;
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->next_ts = timestamp;
self->next_off = GST_BUFFER_OFFSET (inbuf);
} else if (!GST_CLOCK_TIME_IS_VALID (self->next_ts)) {
/* 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_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;
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) -=
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_BUFFER_OFFSET_END (outbuf), output_samples / channels);
return GST_FLOW_OK;
gst_audio_fx_base_fir_filter_start (GstBaseTransform * base)
gst_audio_fx_base_fir_filter_stop (GstBaseTransform * base)
g_free (self->residue);
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_query_set_latency (query, live, min, max);
}
gst_object_unref (peer);
}
break;
}
default:
res = gst_pad_query_default (pad, query);
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;
gst_audio_fx_base_fir_filter_event (GstBaseTransform * base, GstEvent * event)
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_EOS:
self->next_ts = GST_CLOCK_TIME_NONE;
self->next_off = GST_BUFFER_OFFSET_NONE;
return GST_BASE_TRANSFORM_CLASS (parent_class)->event (base, event);
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);
g_free (self->kernel);
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);
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);
--- NEW FILE: audiofxbasefirfilter.h ---
* 2009 Sebastian Dröge <[email protected]>
#ifndef __GST_AUDIO_FX_BASE_FIR_FILTER_H__
#define __GST_AUDIO_FX_BASE_FIR_FILTER_H__
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__ */
Index: audiofxbaseiirfilter.c
RCS file: /cvs/gstreamer/gst-plugins-good/gst/audiofx/audiofxbaseiirfilter.c,v
retrieving revision 1.1
retrieving revision 1.2
diff -u -d -r1.1 -r1.2
--- audiofxbaseiirfilter.c 5 Jan 2009 10:13:28 -0000 1.1
+++ audiofxbaseiirfilter.c 11 Jan 2009 19:03:37 -0000 1.2
@@ -43,7 +43,7 @@
" 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,
Index: audiowsincband.c
RCS file: /cvs/gstreamer/gst-plugins-good/gst/audiofx/audiowsincband.c,v
retrieving revision 1.40
retrieving revision 1.41
diff -u -d -r1.40 -r1.41
--- audiowsincband.c 10 Nov 2008 13:59:26 -0000 1.40
+++ audiowsincband.c 11 Jan 2009 19:03:38 -0000 1.41
@@ -3,7 +3,7 @@
* GStreamer
* Copyright (C) 1999-2001 Erik Walthinsen <[email protected]>
* 2006 Dreamlab Technologies Ltd. <[email protected]>
- * 2007 Sebastian Dröge <[email protected]>
+ * 2007-2009 Sebastian Dröge <[email protected]>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
@@ -74,25 +74,9 @@
#include "audiowsincband.h"
-#define GST_CAT_DEFAULT gst_audio_wsincband_debug
+#define GST_CAT_DEFAULT gst_gst_audio_wsincband_debug
GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
-static const GstElementDetails audio_wsincband_details =
-GST_ELEMENT_DETAILS ("Band pass & band reject filter",
- "Filter/Effect/Audio",
- "Band pass and band reject windowed sinc filter",
- "Thomas Vander Stichele <thomas at apestaart dot org>, "
- "Steven W. Smith, "
- "Dreamlab Technologies Ltd. <[email protected]>, "
- "Sebastian Dröge <[email protected]>");
-
-/* Filter signals and args */
-enum
-{
- /* FILL ME */
- LAST_SIGNAL
-};
enum
{
PROP_0,
@@ -109,9 +93,9 @@
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 @@
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 ())
-gst_audio_wsincband_window_get_type (void)
+gst_gst_audio_wsincband_window_get_type (void)
@@ -155,193 +139,96 @@
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 ] "
- GST_DEBUG_CATEGORY_INIT (gst_audio_wsincband_debug, "audiowsincband", 0, \
+ GST_DEBUG_CATEGORY_INIT (gst_gst_audio_wsincband_debug, "audiowsincband", 0, \
"Band-pass and Band-reject Windowed sinc filter plugin");
-GST_BOILERPLATE_FULL (GstAudioWSincBand, audio_wsincband, GstAudioFilter,
- GST_TYPE_AUDIO_FILTER, DEBUG_INIT);
+GST_BOILERPLATE_FULL (GstAudioWSincBand, gst_audio_wsincband, GstAudioFilter,
+ GST_TYPE_AUDIO_FX_BASE_FIR_FILTER, DEBUG_INIT);
-static void audio_wsincband_set_property (GObject * object, guint prop_id,
+static void gst_audio_wsincband_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
-static void audio_wsincband_get_property (GObject * object, guint prop_id,
+static void gst_audio_wsincband_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
-static GstFlowReturn audio_wsincband_transform (GstBaseTransform * base,
- GstBuffer * inbuf, GstBuffer * outbuf);
-static gboolean audio_wsincband_start (GstBaseTransform * base);
-static gboolean audio_wsincband_event (GstBaseTransform * base,
- GstEvent * event);
-static gboolean audio_wsincband_setup (GstAudioFilter * base,
+static gboolean gst_audio_wsincband_setup (GstAudioFilter * base,
GstRingBufferSpec * format);
-static gboolean audio_wsincband_query (GstPad * pad, GstQuery * query);
-static const GstQueryType *audio_wsincband_query_type (GstPad * pad);
/* Element class */
static void
-audio_wsincband_dispose (GObject * object)
- GstAudioWSincBand *self = GST_AUDIO_WSINC_BAND (object);
- if (self->residue) {
- g_free (self->residue);
- self->residue = NULL;
- }
- if (self->kernel) {
- g_free (self->kernel);
- self->kernel = NULL;
- G_OBJECT_CLASS (parent_class)->dispose (object);
-}
-static void
-audio_wsincband_base_init (gpointer g_class)
+gst_audio_wsincband_base_init (gpointer g_class)
GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
- GstCaps *caps;
- gst_element_class_set_details (element_class, &audio_wsincband_details);
- caps = gst_caps_from_string (ALLOWED_CAPS);
- gst_audio_filter_class_add_pad_templates (GST_AUDIO_FILTER_CLASS (g_class),
- caps);
- gst_caps_unref (caps);
+ gst_element_class_set_details_simple (element_class,
+ "Band pass & band reject filter", "Filter/Effect/Audio",
+ "Band pass and band reject windowed sinc filter",
+ "Thomas Vander Stichele <thomas at apestaart dot org>, "
+ "Steven W. Smith, "
+ "Dreamlab Technologies Ltd. <[email protected]>, "
+ "Sebastian Dröge <[email protected]>");
-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_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_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_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,
- 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);
-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
-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 @@
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 @@
kernel_hp[len / 2] += 1;
/* combine the two kernels */
- if (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 @@
* 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;
-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);
- /* 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);
- 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;
- ret = FALSE;
- return TRUE;
-/* GstBaseTransform vmethod implementations */
-static GstFlowReturn
-audio_wsincband_transform (GstBaseTransform * base, GstBuffer * inbuf,
- GstBuffer * outbuf)
- GstClockTime timestamp;
- 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->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 */
- 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) -=
- } 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;
- 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;
- 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_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)
- /* Reset the residue if already existing */
- if (channels && self->residue)
- memset (self->residue, 0, channels *
- self->kernel_length * sizeof (gdouble));
-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_query_set_latency (query, live, min, max);
- }
- gst_object_unref (peer);
- break;
- default:
- res = gst_pad_query_default (pad, query);
- 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;
-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);
+ 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);
-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 @@
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:
self->lower_frequency = g_value_get_float (value);
- audio_wsincband_build_kernel (self);
+ gst_audio_wsincband_build_kernel (self);
case PROP_UPPER_FREQUENCY:
self->upper_frequency = g_value_get_float (value);
case PROP_MODE:
self->mode = g_value_get_enum (value);
case PROP_WINDOW:
self->window = g_value_get_enum (value);
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
@@ -839,8 +409,8 @@
-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)
Index: audiowsincband.h
RCS file: /cvs/gstreamer/gst-plugins-good/gst/audiofx/audiowsincband.h,v
retrieving revision 1.8
retrieving revision 1.9
diff -u -d -r1.8 -r1.9
--- audiowsincband.h 7 Feb 2008 21:57:54 -0000 1.8
+++ audiowsincband.h 11 Jan 2009 19:03:38 -0000 1.9
@@ -3,6 +3,7 @@
@@ -33,10 +34,12 @@
#include <gst/gst.h>
#include <gst/audio/gstaudiofilter.h>
+#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 @@
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
Index: audiowsinclimit.c
RCS file: /cvs/gstreamer/gst-plugins-good/gst/audiofx/audiowsinclimit.c,v
retrieving revision 1.46
retrieving revision 1.47
diff -u -d -r1.46 -r1.47
--- audiowsinclimit.c 10 Nov 2008 13:59:26 -0000 1.46
+++ audiowsinclimit.c 11 Jan 2009 19:03:38 -0000 1.47
@@ -72,25 +72,9 @@
#include "audiowsinclimit.h"
-#define GST_CAT_DEFAULT audio_wsinclimit_debug
+#define GST_CAT_DEFAULT gst_audio_wsinclimit_debug
-static const GstElementDetails audio_wsinclimit_details =
-GST_ELEMENT_DETAILS ("Low pass & high pass filter",
- "Low pass and high pass windowed sinc filter",
@@ -106,9 +90,9 @@
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 ())
-audio_wsinclimit_mode_get_type (void)
+gst_audio_wsinclimit_mode_get_type (void)
@@ -132,9 +116,9 @@
-#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 ())
-audio_wsinclimit_window_get_type (void)
+gst_audio_wsinclimit_window_get_type (void)
@@ -152,189 +136,91 @@
- " channels = (int) [ 1, MAX ]"
- 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_BOILERPLATE_FULL (GstAudioWSincLimit, gst_audio_wsinclimit, GstAudioFilter,
-static void audio_wsinclimit_set_property (GObject * object, guint prop_id,
+static void gst_audio_wsinclimit_set_property (GObject * object, guint prop_id,
-static void audio_wsinclimit_get_property (GObject * object, guint prop_id,
+static void gst_audio_wsinclimit_get_property (GObject * object, guint prop_id,
-static GstFlowReturn audio_wsinclimit_transform (GstBaseTransform * base,
-static gboolean audio_wsinclimit_start (GstBaseTransform * base);
-static gboolean audio_wsinclimit_event (GstBaseTransform * base,
-static gboolean audio_wsinclimit_setup (GstAudioFilter * base,
+static gboolean gst_audio_wsinclimit_setup (GstAudioFilter * base,
-static gboolean audio_wsinclimit_query (GstPad * pad, GstQuery * query);
-static const GstQueryType *audio_wsinclimit_query_type (GstPad * pad);
-audio_wsinclimit_dispose (GObject * object)
- GstAudioWSincLimit *self = GST_AUDIO_WSINC_LIMIT (object);
-audio_wsinclimit_base_init (gpointer g_class)
+gst_audio_wsinclimit_base_init (gpointer g_class)
- gst_element_class_set_details (element_class, &audio_wsinclimit_details);
+ "Low pass & high pass filter", "Filter/Effect/Audio",
+ "Low pass and high pass windowed sinc filter",
-audio_wsinclimit_class_init (GstAudioWSincLimitClass * klass)
+gst_audio_wsinclimit_class_init (GstAudioWSincLimitClass * klass)
- gobject_class->set_property = audio_wsinclimit_set_property;
- gobject_class->get_property = audio_wsinclimit_get_property;
- gobject_class->dispose = audio_wsinclimit_dispose;
+ gobject_class->set_property = gst_audio_wsinclimit_set_property;
+ gobject_class->get_property = gst_audio_wsinclimit_get_property;
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));
"Filter kernel length, will be rounded to the next odd number",
- 3, 50000, 101, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
+ 3, 50000, 101,
"Low pass or high pass mode", GST_TYPE_AUDIO_WSINC_LIMIT_MODE,
- MODE_LOW_PASS, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
+ MODE_LOW_PASS,
"Window function to use", GST_TYPE_AUDIO_WSINC_LIMIT_WINDOW,
- 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);
-audio_wsinclimit_init (GstAudioWSincLimit * self,
+gst_audio_wsinclimit_init (GstAudioWSincLimit * self,
GstAudioWSincLimitClass * g_class)
self->mode = MODE_LOW_PASS;
self->cutoff = 0.0;
- audio_wsinclimit_query);
- audio_wsinclimit_query_type);
-process_##width (GstAudioWSincLimit * self, g##ctype * src, g##ctype * dst, guint input_samples) \
-audio_wsinclimit_build_kernel (GstAudioWSincLimit * self)
+gst_audio_wsinclimit_build_kernel (GstAudioWSincLimit * self)
+ gdouble *kernel = NULL;
@@ -352,7 +238,7 @@
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 @@
/* fill the kernel */
w = 2 * M_PI * (self->cutoff / GST_AUDIO_FILTER (self)->format.rate);
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));
- 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 */
- sum += self->kernel[i];
+ sum += kernel[i];
- self->kernel[i] /= sum;
+ kernel[i] /= sum;
/* convert to highpass if specified */
if (self->mode == MODE_HIGH_PASS) {
- self->kernel[len / 2] += 1.0;
-audio_wsinclimit_push_residue (GstAudioWSincLimit * self)
+ kernel[len / 2] += 1.0;
-audio_wsinclimit_setup (GstAudioFilter * base, GstRingBufferSpec * format)
- GstAudioWSincLimit *self = GST_AUDIO_WSINC_LIMIT (base);
- self->process = (GstAudioWSincLimitProcessFunc) process_32;
- self->process = (GstAudioWSincLimitProcessFunc) process_64;
-audio_wsinclimit_transform (GstBaseTransform * base, GstBuffer * inbuf,
- audio_wsinclimit_build_kernel (self);
-audio_wsinclimit_start (GstBaseTransform * base)
-audio_wsinclimit_query (GstPad * pad, GstQuery * query)
- GstAudioWSincLimit *self = GST_AUDIO_WSINC_LIMIT (gst_pad_get_parent (pad));
-audio_wsinclimit_query_type (GstPad * pad)
-audio_wsinclimit_event (GstBaseTransform * base, GstEvent * event)
+gst_audio_wsinclimit_setup (GstAudioFilter * base, GstRingBufferSpec * format)
GstAudioWSincLimit *self = GST_AUDIO_WSINC_LIMIT (base);
- audio_wsinclimit_push_residue (self);
+ gst_audio_wsinclimit_build_kernel (self);
-audio_wsinclimit_set_property (GObject * object, guint prop_id,
+gst_audio_wsinclimit_set_property (GObject * object, guint prop_id,
GstAudioWSincLimit *self = GST_AUDIO_WSINC_LIMIT (object);
@@ -730,43 +304,37 @@
- audio_wsinclimit_push_residue (self);
- audio_wsinclimit_build_kernel (self);
+ gst_audio_wsinclimit_build_kernel (self);
case PROP_FREQUENCY:
self->cutoff = g_value_get_float (value);
- audio_wsinclimit_build_kernel (self);
+ gst_audio_wsinclimit_build_kernel (self);
@@ -775,8 +343,8 @@
-audio_wsinclimit_get_property (GObject * object, guint prop_id, GValue * value,
+gst_audio_wsinclimit_get_property (GObject * object, guint prop_id,
Index: audiowsinclimit.h
RCS file: /cvs/gstreamer/gst-plugins-good/gst/audiofx/audiowsinclimit.h,v
retrieving revision 1.10
retrieving revision 1.11
diff -u -d -r1.10 -r1.11
--- audiowsinclimit.h 7 Feb 2008 21:57:54 -0000 1.10
+++ audiowsinclimit.h 11 Jan 2009 19:03:38 -0000 1.11
#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) \
typedef struct _GstAudioWSincLimit GstAudioWSincLimit;
typedef struct _GstAudioWSincLimitClass GstAudioWSincLimitClass;
-typedef void (*GstAudioWSincLimitProcessFunc) (GstAudioWSincLimit *, guint8 *, guint8 *, guint);
* GstAudioWSincLimit:
struct _GstAudioWSincLimit {
- GstAudioWSincLimitProcessFunc process;
gfloat cutoff;
- gint kernel_length; /* length of the filter kernel */
- gdouble *kernel; /* filter kernel */
+ gint kernel_length;
struct _GstAudioWSincLimitClass {
+ GstAudioFXBaseFIRFilterClass parent;
-GType audio_wsinclimit_get_type (void);
+GType gst_audio_wsinclimit_get_type (void);
RCS file: /cvs/gstreamer/gst-plugins-good/tests/check/elements/audiowsincband.c,v
retrieving revision 1.5
retrieving revision 1.6
diff -u -d -r1.5 -r1.6
--- audiowsincband.c 7 Feb 2008 21:57:54 -0000 1.5
+++ audiowsincband.c 11 Jan 2009 19:03:38 -0000 1.6
@@ -119,6 +119,7 @@
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 @@
for (i = 0; i < 1024; i += 4) {
in[i] = 0.0;
@@ -246,6 +248,7 @@
for (i = 0; i < 1024; i += 2) {
@@ -309,6 +312,7 @@
@@ -370,6 +374,7 @@
@@ -437,6 +442,7 @@
@@ -498,6 +504,7 @@
44100 / 4.0 + 44100 / 16.0, NULL);
inbuffer = gst_buffer_new_and_alloc (20 * sizeof (gfloat));
for (i = 0; i < 20; i++)
@@ -553,6 +560,7 @@
inbuffer = gst_buffer_new_and_alloc (1024 * sizeof (gdouble));
in = (gdouble *) GST_BUFFER_DATA (inbuffer);
@@ -614,6 +622,7 @@
@@ -680,6 +689,7 @@
@@ -743,6 +753,7 @@
@@ -804,6 +815,7 @@
@@ -871,6 +883,7 @@
@@ -932,6 +945,7 @@
inbuffer = gst_buffer_new_and_alloc (20 * sizeof (gdouble));
RCS file: /cvs/gstreamer/gst-plugins-good/tests/check/elements/audiowsinclimit.c,v
--- audiowsinclimit.c 7 Feb 2008 21:57:54 -0000 1.5
+++ audiowsinclimit.c 11 Jan 2009 19:03:38 -0000 1.6
@@ -117,6 +117,7 @@
/* 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));
for (i = 0; i < 128; i++)
@@ -175,6 +176,7 @@
for (i = 0; i < 128; i += 2) {
@@ -235,6 +237,7 @@
@@ -293,6 +296,7 @@
@@ -352,6 +356,7 @@
@@ -398,6 +403,7 @@
inbuffer = gst_buffer_new_and_alloc (128 * sizeof (gdouble));
@@ -456,6 +462,7 @@
@@ -516,6 +523,7 @@
@@ -574,6 +582,7 @@
@@ -633,6 +642,7 @@
--===============4753564101550686637==
Content-Type: text/plain; charset="us-ascii"
MIME-Version: 1.0
Content-Transfer-Encoding: 7bit
Content-Disposition: inline
------------------------------------------------------------------------------
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--===============4753564101550686637==
Content-Type: text/plain; charset="us-ascii"
MIME-Version: 1.0
Content-Transfer-Encoding: 7bit
Content-Disposition: inline
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--===============4753564101550686637==--