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gst/audiorate/: Added an audiorate converter that fills in gaps.
Original commit message from CVS: * gst/audiorate/Makefile.am: * gst/audiorate/gstaudiorate.c: (gst_audiorate_get_type), (gst_audiorate_base_init), (gst_audiorate_class_init), (gst_audiorate_link), (gst_audiorate_init), (gst_audiorate_chain), (gst_audiorate_set_property), (gst_audiorate_get_property), (gst_audiorate_change_state), (plugin_init): Added an audiorate converter that fills in gaps.
This commit is contained in:
Wim Taymans
parent
c163b09cc4
commit
7face56b42
4 changed files with 377 additions and 0 deletions
10
ChangeLog
10
ChangeLog
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@ -1,3 +1,13 @@
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2004-06-17 Wim Taymans <wim@fluendo.com>
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* gst/audiorate/Makefile.am:
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* gst/audiorate/gstaudiorate.c: (gst_audiorate_get_type),
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(gst_audiorate_base_init), (gst_audiorate_class_init),
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(gst_audiorate_link), (gst_audiorate_init), (gst_audiorate_chain),
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(gst_audiorate_set_property), (gst_audiorate_get_property),
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(gst_audiorate_change_state), (plugin_init):
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Added an audiorate converter that fills in gaps.
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2004-06-17 Johan Dahlin <johan@gnome.org>
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2004-06-17 Johan Dahlin <johan@gnome.org>
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* ext/tcp/*: Revert Zaheer changes.
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* ext/tcp/*: Revert Zaheer changes.
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@ -340,6 +340,7 @@ GST_PLUGINS_ALL="\
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asfdemux \
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asfdemux \
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audioconvert \
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audioconvert \
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audioscale \
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audioscale \
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audiorate \
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auparse \
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auparse \
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avi \
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avi \
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cdxaparse \
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cdxaparse \
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@ -395,6 +396,7 @@ GST_PLUGINS_ALL="\
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videoflip \
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videoflip \
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videofilter \
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videofilter \
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videomixer \
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videomixer \
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videorate \
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videoscale \
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videoscale \
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videotestsrc \
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videotestsrc \
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volenv \
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volenv \
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@ -1781,6 +1783,7 @@ gst/adder/Makefile
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gst/alpha/Makefile
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gst/alpha/Makefile
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gst/audioconvert/Makefile
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gst/audioconvert/Makefile
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gst/audioscale/Makefile
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gst/audioscale/Makefile
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gst/audiorate/Makefile
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gst/auparse/Makefile
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gst/auparse/Makefile
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gst/avi/Makefile
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gst/avi/Makefile
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gst/asfdemux/Makefile
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gst/asfdemux/Makefile
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@ -1840,6 +1843,7 @@ gst/videofilter/Makefile
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gst/videoflip/Makefile
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gst/videoflip/Makefile
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gst/videomixer/Makefile
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gst/videomixer/Makefile
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gst/videoscale/Makefile
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gst/videoscale/Makefile
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gst/videorate/Makefile
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gst/videotestsrc/Makefile
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gst/videotestsrc/Makefile
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gst/volenv/Makefile
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gst/volenv/Makefile
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gst/volume/Makefile
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gst/volume/Makefile
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7
gst/audiorate/Makefile.am
Normal file
7
gst/audiorate/Makefile.am
Normal file
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@ -0,0 +1,7 @@
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plugin_LTLIBRARIES = libgstaudiorate.la
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libgstaudiorate_la_SOURCES = gstaudiorate.c
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libgstaudiorate_la_CFLAGS = $(GST_CFLAGS)
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libgstaudiorate_la_LIBADD =
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libgstaudiorate_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
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356
gst/audiorate/gstaudiorate.c
Normal file
356
gst/audiorate/gstaudiorate.c
Normal file
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@ -0,0 +1,356 @@
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/* GStreamer
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* Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
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* Boston, MA 02111-1307, USA.
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*/
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#ifdef HAVE_CONFIG_H
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#include "config.h"
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#endif
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#include <string.h>
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#include <gst/gst.h>
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#include <gst/audio/audio.h>
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#define GST_TYPE_AUDIORATE \
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(gst_audiorate_get_type())
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#define GST_AUDIORATE(obj) \
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(G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIORATE,GstAudiorate))
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#define GST_AUDIORATE_CLASS(klass) \
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(G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_AUDIORATE,GstAudiorate))
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#define GST_IS_AUDIORATE(obj) \
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(G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIORATE))
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#define GST_IS_AUDIORATE_CLASS(obj) \
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(G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_AUDIORATE))
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typedef struct _GstAudiorate GstAudiorate;
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typedef struct _GstAudiorateClass GstAudiorateClass;
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struct _GstAudiorate
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{
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GstElement element;
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GstPad *sinkpad, *srcpad;
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/* audio state */
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guint64 next_offset;
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guint64 in, out, add, drop;
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};
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struct _GstAudiorateClass
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{
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GstElementClass parent_class;
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};
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/* elementfactory information */
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static GstElementDetails audiorate_details =
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GST_ELEMENT_DETAILS ("Audio rate adjuster",
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"Filter/Effect/Audio",
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"Drops/duplicates/adjusts timestamps on audio samples to make a perfect stream",
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"Wim Taymans <wim@fluendo.com>");
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/* GstAudiorate signals and args */
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enum
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{
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/* FILL ME */
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LAST_SIGNAL
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};
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enum
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{
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ARG_0,
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ARG_IN,
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ARG_OUT,
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ARG_ADD,
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ARG_DROP,
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/* FILL ME */
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};
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static GstStaticPadTemplate gst_audiorate_src_template =
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GST_STATIC_PAD_TEMPLATE ("src",
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GST_PAD_SRC,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS (GST_AUDIO_INT_PAD_TEMPLATE_CAPS)
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);
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static GstStaticPadTemplate gst_audiorate_sink_template =
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GST_STATIC_PAD_TEMPLATE ("sink",
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GST_PAD_SINK,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS (GST_AUDIO_INT_PAD_TEMPLATE_CAPS)
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);
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static void gst_audiorate_base_init (gpointer g_class);
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static void gst_audiorate_class_init (GstAudiorateClass * klass);
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static void gst_audiorate_init (GstAudiorate * audiorate);
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static void gst_audiorate_chain (GstPad * pad, GstData * _data);
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static void gst_audiorate_set_property (GObject * object,
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guint prop_id, const GValue * value, GParamSpec * pspec);
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static void gst_audiorate_get_property (GObject * object,
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guint prop_id, GValue * value, GParamSpec * pspec);
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static GstElementStateReturn gst_audiorate_change_state (GstElement * element);
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static GstElementClass *parent_class = NULL;
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/*static guint gst_audiorate_signals[LAST_SIGNAL] = { 0 }; */
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static GType
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gst_audiorate_get_type (void)
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{
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static GType audiorate_type = 0;
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if (!audiorate_type) {
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static const GTypeInfo audiorate_info = {
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sizeof (GstAudiorateClass),
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gst_audiorate_base_init,
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NULL,
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(GClassInitFunc) gst_audiorate_class_init,
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NULL,
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NULL,
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sizeof (GstAudiorate),
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0,
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(GInstanceInitFunc) gst_audiorate_init,
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};
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audiorate_type = g_type_register_static (GST_TYPE_ELEMENT,
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"GstAudiorate", &audiorate_info, 0);
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}
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return audiorate_type;
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}
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static void
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gst_audiorate_base_init (gpointer g_class)
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{
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GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
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gst_element_class_set_details (element_class, &audiorate_details);
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&gst_audiorate_sink_template));
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&gst_audiorate_src_template));
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}
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static void
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gst_audiorate_class_init (GstAudiorateClass * klass)
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{
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GObjectClass *object_class = G_OBJECT_CLASS (klass);
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GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
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parent_class = g_type_class_peek_parent (klass);
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g_object_class_install_property (object_class, ARG_IN,
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g_param_spec_uint64 ("in", "In",
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"Number of input samples", 0, G_MAXUINT64, 0, G_PARAM_READABLE));
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g_object_class_install_property (object_class, ARG_OUT,
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g_param_spec_uint64 ("out", "Out",
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"Number of output samples", 0, G_MAXUINT64, 0, G_PARAM_READABLE));
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g_object_class_install_property (object_class, ARG_ADD,
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g_param_spec_uint64 ("duplicate", "Duplicate",
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"Number of added samples", 0, G_MAXUINT64, 0, G_PARAM_READABLE));
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g_object_class_install_property (object_class, ARG_DROP,
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g_param_spec_uint64 ("drop", "Drop",
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"Number of dropped samples", 0, G_MAXUINT64, 0, G_PARAM_READABLE));
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object_class->set_property = gst_audiorate_set_property;
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object_class->get_property = gst_audiorate_get_property;
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element_class->change_state = gst_audiorate_change_state;
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}
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static GstPadLinkReturn
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gst_audiorate_link (GstPad * pad, const GstCaps * caps)
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{
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GstAudiorate *audiorate;
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GstStructure *structure;
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GstPad *otherpad;
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GstPadLinkReturn res;
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audiorate = GST_AUDIORATE (gst_pad_get_parent (pad));
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otherpad = (pad == audiorate->srcpad) ? audiorate->sinkpad :
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audiorate->srcpad;
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res = gst_pad_try_set_caps (otherpad, caps);
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if (GST_PAD_LINK_FAILED (res))
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return res;
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structure = gst_caps_get_structure (caps, 0);
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return GST_PAD_LINK_OK;
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}
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static void
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gst_audiorate_init (GstAudiorate * audiorate)
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{
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GST_FLAG_SET (audiorate, GST_ELEMENT_EVENT_AWARE);
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GST_DEBUG ("gst_audiorate_init");
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audiorate->sinkpad =
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gst_pad_new_from_template (gst_static_pad_template_get
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(&gst_audiorate_sink_template), "sink");
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gst_element_add_pad (GST_ELEMENT (audiorate), audiorate->sinkpad);
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gst_pad_set_chain_function (audiorate->sinkpad, gst_audiorate_chain);
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gst_pad_set_link_function (audiorate->sinkpad, gst_audiorate_link);
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gst_pad_set_getcaps_function (audiorate->sinkpad, gst_pad_proxy_getcaps);
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audiorate->srcpad =
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gst_pad_new_from_template (gst_static_pad_template_get
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(&gst_audiorate_src_template), "src");
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gst_element_add_pad (GST_ELEMENT (audiorate), audiorate->srcpad);
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gst_pad_set_link_function (audiorate->srcpad, gst_audiorate_link);
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gst_pad_set_getcaps_function (audiorate->srcpad, gst_pad_proxy_getcaps);
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audiorate->in = 0;
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audiorate->out = 0;
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audiorate->drop = 0;
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audiorate->add = 0;
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}
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static void
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gst_audiorate_chain (GstPad * pad, GstData * data)
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{
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GstAudiorate *audiorate;
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GstBuffer *buf;
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GstClockTime in_time, in_duration;
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guint64 in_offset, in_offset_end;
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gint in_size;
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audiorate = GST_AUDIORATE (gst_pad_get_parent (pad));
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if (GST_IS_EVENT (data)) {
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GstEvent *event = GST_EVENT (data);
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gst_pad_event_default (pad, event);
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return;
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}
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audiorate->in++;
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buf = GST_BUFFER (data);
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in_time = GST_BUFFER_TIMESTAMP (buf);
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in_duration = GST_BUFFER_DURATION (buf);
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in_size = GST_BUFFER_SIZE (buf);
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in_offset = GST_BUFFER_OFFSET (buf);
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in_offset_end = GST_BUFFER_OFFSET_END (buf);
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/* do we need to insert samples */
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if (in_offset > audiorate->next_offset) {
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GstBuffer *fill;
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gint bytes_per_sample, fillsize;
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guint64 fillsamples;
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/* FIXME: use caps to get this */
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bytes_per_sample = in_size / (in_offset_end - in_offset);
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fillsamples = in_offset - audiorate->next_offset;
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fillsize = fillsamples * bytes_per_sample;
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fill = gst_buffer_new_and_alloc (fillsize);
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memset (GST_BUFFER_DATA (fill), 0, fillsize);
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GST_BUFFER_DURATION (fill) = in_duration * fillsize / in_size;
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GST_BUFFER_TIMESTAMP (fill) = in_time - GST_BUFFER_DURATION (fill);
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GST_BUFFER_OFFSET (fill) = audiorate->next_offset;
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GST_BUFFER_OFFSET_END (fill) = in_offset;
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gst_pad_push (audiorate->srcpad, GST_DATA (fill));
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audiorate->out++;
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audiorate->add++;
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} else if (in_offset < audiorate->next_offset) {
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g_warning ("overlapping samples, implement me");
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audiorate->drop++;
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}
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||||||
|
gst_pad_push (audiorate->srcpad, GST_DATA (buf));
|
||||||
|
audiorate->out++;
|
||||||
|
|
||||||
|
audiorate->next_offset = in_offset_end;
|
||||||
|
}
|
||||||
|
|
||||||
|
static void
|
||||||
|
gst_audiorate_set_property (GObject * object,
|
||||||
|
guint prop_id, const GValue * value, GParamSpec * pspec)
|
||||||
|
{
|
||||||
|
//GstAudiorate *audiorate = GST_AUDIORATE (object);
|
||||||
|
|
||||||
|
switch (prop_id) {
|
||||||
|
default:
|
||||||
|
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
|
||||||
|
break;
|
||||||
|
}
|
||||||
|
}
|
||||||
|
|
||||||
|
static void
|
||||||
|
gst_audiorate_get_property (GObject * object,
|
||||||
|
guint prop_id, GValue * value, GParamSpec * pspec)
|
||||||
|
{
|
||||||
|
GstAudiorate *audiorate = GST_AUDIORATE (object);
|
||||||
|
|
||||||
|
switch (prop_id) {
|
||||||
|
case ARG_IN:
|
||||||
|
g_value_set_uint64 (value, audiorate->in);
|
||||||
|
break;
|
||||||
|
case ARG_OUT:
|
||||||
|
g_value_set_uint64 (value, audiorate->out);
|
||||||
|
break;
|
||||||
|
case ARG_ADD:
|
||||||
|
g_value_set_uint64 (value, audiorate->add);
|
||||||
|
break;
|
||||||
|
case ARG_DROP:
|
||||||
|
g_value_set_uint64 (value, audiorate->drop);
|
||||||
|
break;
|
||||||
|
default:
|
||||||
|
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
|
||||||
|
break;
|
||||||
|
}
|
||||||
|
}
|
||||||
|
|
||||||
|
static GstElementStateReturn
|
||||||
|
gst_audiorate_change_state (GstElement * element)
|
||||||
|
{
|
||||||
|
GstAudiorate *audiorate = GST_AUDIORATE (element);
|
||||||
|
|
||||||
|
switch (GST_STATE_TRANSITION (element)) {
|
||||||
|
case GST_STATE_PAUSED_TO_READY:
|
||||||
|
break;
|
||||||
|
case GST_STATE_READY_TO_PAUSED:
|
||||||
|
audiorate->next_offset = 0;
|
||||||
|
break;
|
||||||
|
default:
|
||||||
|
break;
|
||||||
|
}
|
||||||
|
|
||||||
|
if (parent_class->change_state)
|
||||||
|
return parent_class->change_state (element);
|
||||||
|
|
||||||
|
return GST_STATE_SUCCESS;
|
||||||
|
}
|
||||||
|
|
||||||
|
static gboolean
|
||||||
|
plugin_init (GstPlugin * plugin)
|
||||||
|
{
|
||||||
|
return gst_element_register (plugin, "audiorate", GST_RANK_NONE,
|
||||||
|
GST_TYPE_AUDIORATE);
|
||||||
|
}
|
||||||
|
|
||||||
|
GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
|
||||||
|
GST_VERSION_MINOR,
|
||||||
|
"audiorate",
|
||||||
|
"Adjusts audio frames",
|
||||||
|
plugin_init, VERSION, GST_LICENSE, GST_PACKAGE, GST_ORIGIN)
|
||||||
Loading…
Reference in a new issue