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port audioresample to basetransform
Original commit message from CVS: port audioresample to basetransform
This commit is contained in:
Thomas Vander Stichele
parent
41a43b86a8
commit
752a59192c
17
ChangeLog
17
ChangeLog
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@ -1,3 +1,20 @@
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2005-08-24 Thomas Vander Stichele <thomas at apestaart dot org>
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* configure.ac:
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compile audioresample
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* gst/audioresample/Makefile.am:
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* gst/audioresample/buffer.c:
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* gst/audioresample/functable.c:
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* gst/audioresample/gstaudioresample.c:
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* gst/audioresample/gstaudioresample.h:
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* gst/audioresample/resample.c:
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(resample_get_output_size_for_input):
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* gst/audioresample/resample.h:
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* gst/audioresample/resample_chunk.c:
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* gst/audioresample/resample_functable.c:
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* gst/audioresample/resample_ref.c:
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port to use basetransform; doesn't work in all cases yet
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2005-08-24 Thomas Vander Stichele <thomas at apestaart dot org>
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* gst/audioconvert/gstaudioconvert.c:
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@ -369,8 +369,8 @@ dnl these are all the gst plug-ins, compilable without additional libs
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GST_PLUGINS_ALL="\
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adder \
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audioconvert \
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audioscale \
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audiorate \
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audioresample \
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ffmpegcolorspace \
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playback \
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sine \
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@ -892,6 +892,7 @@ gst/adder/Makefile
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gst/audioconvert/Makefile
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gst/audioscale/Makefile
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gst/audiorate/Makefile
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gst/audioresample/Makefile
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gst/ffmpegcolorspace/Makefile
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gst/playback/Makefile
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gst/sine/Makefile
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@ -15,7 +15,7 @@ resample_SOURCES = \
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buffer.h
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libgstaudioresample_la_SOURCES = gstaudioresample.c $(resample_SOURCES)
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libgstaudioresample_la_CFLAGS = $(GST_CFLAGS) $(LIBOIL_CFLAGS)
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libgstaudioresample_la_LIBADD = $(LIBOIL_LIBS)
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libgstaudioresample_la_CFLAGS = $(GST_BASE_CFLAGS) $(GST_CFLAGS) $(LIBOIL_CFLAGS)
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libgstaudioresample_la_LIBADD = $(GST_BASE_LIBS) $(GST_LIBS) $(LIBOIL_LIBS)
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libgstaudioresample_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
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@ -3,10 +3,11 @@
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#include "config.h"
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#endif
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#include <audioresample/buffer.h>
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#include <glib.h>
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#include <string.h>
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#include <audioresample/debug.h>
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#include "buffer.h"
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#include "debug.h"
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static void audioresample_buffer_free_mem (AudioresampleBuffer * buffer,
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void *);
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@ -26,8 +26,8 @@
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#include <stdio.h>
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#include <stdlib.h>
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#include <audioresample/functable.h>
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#include <audioresample/debug.h>
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#include "functable.h"
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#include "debug.h"
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@ -1,5 +1,5 @@
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/* GStreamer
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* Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
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* Copyright (C) 1999 Erik Walthinsen <omega@cse.ogi.edu>
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* Copyright (C) 2003,2004 David A. Schleef <ds@schleef.org>
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*
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* This library is free software; you can redistribute it and/or
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@ -19,16 +19,17 @@
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*/
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/* Element-Checklist-Version: 5 */
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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 <math.h>
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/*#define DEBUG_ENABLED */
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#include "gstaudioresample.h"
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#include <gst/audio/audio.h>
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#include <gst/base/gstbasetransform.h>
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GST_DEBUG_CATEGORY_STATIC (audioresample_debug);
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#define GST_CAT_DEFAULT audioresample_debug
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@ -40,7 +41,7 @@ GST_ELEMENT_DETAILS ("Audio scaler",
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"Resample audio",
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"David Schleef <ds@schleef.org>");
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/* Audioresample signals and args */
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/* GstAudioresample signals and args */
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enum
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{
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/* FILL ME */
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@ -79,63 +80,54 @@ enum
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GST_PAD_SRC, GST_PAD_ALWAYS, SUPPORTED_CAPS);
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static void gst_audioresample_base_init (gpointer g_class);
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static void gst_audioresample_class_init (AudioresampleClass * klass);
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static void gst_audioresample_init (Audioresample * audioresample);
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static void gst_audioresample_class_init (GstAudioresampleClass * klass);
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static void gst_audioresample_init (GstAudioresample * audioresample);
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static void gst_audioresample_dispose (GObject * object);
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static void gst_audioresample_chain (GstPad * pad, GstData * _data);
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static void gst_audioresample_set_property (GObject * object,
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guint prop_id, const GValue * value, GParamSpec * pspec);
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static void gst_audioresample_get_property (GObject * object,
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guint prop_id, GValue * value, GParamSpec * pspec);
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static GstElementClass *parent_class = NULL;
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/* vmethods */
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gboolean audioresample_get_unit_size (GstBaseTransform * base,
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GstCaps * caps, guint * size);
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GstCaps *audioresample_transform_caps (GstBaseTransform * base,
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GstPadDirection direction, GstCaps * caps);
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gboolean audioresample_transform_size (GstBaseTransform * trans,
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GstPadDirection direction, GstCaps * incaps, guint insize,
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GstCaps * outcaps, guint * outsize);
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gboolean audioresample_set_caps (GstBaseTransform * base, GstCaps * incaps,
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GstCaps * outcaps);
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static GstFlowReturn audioresample_transform (GstBaseTransform * base,
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GstBuffer * inbuf, GstBuffer * outbuf);
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/*static guint gst_audioresample_signals[LAST_SIGNAL] = { 0 }; */
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GType audioresample_get_type (void)
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#define DEBUG_INIT(bla) \
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GST_DEBUG_CATEGORY_INIT (audioresample_debug, "audioresample", 0, "audio resampling element");
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GST_BOILERPLATE_FULL (GstAudioresample, gst_audioresample, GstBaseTransform,
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GST_TYPE_BASE_TRANSFORM, DEBUG_INIT);
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static void gst_audioresample_base_init (gpointer g_class)
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{
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static GType audioresample_type = 0;
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GstElementClass *gstelement_class = GST_ELEMENT_CLASS (g_class);
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if (!audioresample_type)
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{
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static const GTypeInfo audioresample_info = {
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sizeof (AudioresampleClass),
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gst_audioresample_base_init,
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NULL,
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(GClassInitFunc) gst_audioresample_class_init,
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NULL,
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NULL,
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sizeof (Audioresample), 0,
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(GInstanceInitFunc) gst_audioresample_init,};
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gst_element_class_add_pad_template (gstelement_class,
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gst_static_pad_template_get (&gst_audioresample_src_template));
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gst_element_class_add_pad_template (gstelement_class,
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gst_static_pad_template_get (&gst_audioresample_sink_template));
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audioresample_type =
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g_type_register_static (GST_TYPE_ELEMENT, "Audioresample",
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&audioresample_info, 0);
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}
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return audioresample_type;
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gst_element_class_set_details (gstelement_class,
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&gst_audioresample_details);
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}
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static void gst_audioresample_base_init (gpointer g_class)
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{
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GstElementClass *gstelement_class = GST_ELEMENT_CLASS (g_class);
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gst_element_class_add_pad_template (gstelement_class,
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gst_static_pad_template_get (&gst_audioresample_src_template));
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gst_element_class_add_pad_template (gstelement_class,
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gst_static_pad_template_get (&gst_audioresample_sink_template));
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gst_element_class_set_details (gstelement_class, &gst_audioresample_details);
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}
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static void gst_audioresample_class_init (AudioresampleClass * klass)
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static void gst_audioresample_class_init (GstAudioresampleClass * klass)
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{
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GObjectClass *gobject_class;
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GstElementClass *gstelement_class;
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gobject_class = (GObjectClass *) klass;
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gstelement_class = (GstElementClass *) klass;
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gobject_class->set_property = gst_audioresample_set_property;
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gobject_class->get_property = gst_audioresample_get_property;
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@ -145,163 +137,22 @@ static void gst_audioresample_class_init (AudioresampleClass * klass)
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g_param_spec_int ("filter_length", "filter_length", "filter_length",
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0, G_MAXINT, 16, G_PARAM_READWRITE | G_PARAM_CONSTRUCT));
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parent_class = g_type_class_ref (GST_TYPE_ELEMENT);
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GST_DEBUG_CATEGORY_INIT (audioresample_debug, "audioresample", 0,
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"audioresample element");
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GST_BASE_TRANSFORM_CLASS (klass)->transform_size =
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GST_DEBUG_FUNCPTR (audioresample_transform_size);
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GST_BASE_TRANSFORM_CLASS (klass)->get_unit_size =
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GST_DEBUG_FUNCPTR (audioresample_get_unit_size);
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GST_BASE_TRANSFORM_CLASS (klass)->transform_caps =
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GST_DEBUG_FUNCPTR (audioresample_transform_caps);
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GST_BASE_TRANSFORM_CLASS (klass)->set_caps =
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GST_DEBUG_FUNCPTR (audioresample_set_caps);
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GST_BASE_TRANSFORM_CLASS (klass)->transform =
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GST_DEBUG_FUNCPTR (audioresample_transform);
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}
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static void gst_audioresample_expand_caps (GstCaps * caps)
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{
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gint i;
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for (i = 0; i < gst_caps_get_size (caps); i++) {
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GstStructure *structure = gst_caps_get_structure (caps, i);
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const GValue *value;
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value = gst_structure_get_value (structure, "rate");
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if (value == NULL) {
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GST_ERROR ("caps structure doesn't have required rate field");
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return;
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}
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gst_structure_set (structure, "rate", GST_TYPE_INT_RANGE, 1, G_MAXINT, 0);
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}
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}
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static GstCaps *gst_audioresample_getcaps (GstPad * pad)
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{
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Audioresample *audioresample;
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GstCaps *caps;
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GstPad *otherpad;
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audioresample = GST_AUDIORESAMPLE (gst_pad_get_parent (pad));
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otherpad = (pad == audioresample->srcpad) ? audioresample->sinkpad :
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audioresample->srcpad;
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caps = gst_pad_get_allowed_caps (otherpad);
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gst_audioresample_expand_caps (caps);
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return caps;
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}
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static GstCaps *gst_audioresample_fixate (GstPad * pad, const GstCaps * caps)
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{
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Audioresample *audioresample;
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GstPad *otherpad;
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int rate;
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GstCaps *copy;
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GstStructure *structure;
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audioresample = GST_AUDIORESAMPLE (gst_pad_get_parent (pad));
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if (pad == audioresample->srcpad) {
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otherpad = audioresample->sinkpad;
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rate = audioresample->i_rate;
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} else
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{
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otherpad = audioresample->srcpad;
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rate = audioresample->o_rate;
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}
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if (!GST_PAD_IS_NEGOTIATING (otherpad))
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return NULL;
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if (gst_caps_get_size (caps) > 1)
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return NULL;
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copy = gst_caps_copy (caps);
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structure = gst_caps_get_structure (copy, 0);
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if (rate) {
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if (gst_caps_structure_fixate_field_nearest_int (structure, "rate", rate)) {
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return copy;
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}
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}
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gst_caps_free (copy);
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return NULL;
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}
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static GstPadLinkReturn gst_audioresample_link (GstPad * pad,
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const GstCaps * caps)
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{
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Audioresample *audioresample;
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GstStructure *structure;
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int rate;
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int channels;
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gboolean ret;
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GstPad *otherpad;
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audioresample = GST_AUDIORESAMPLE (gst_pad_get_parent (pad));
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otherpad = (pad == audioresample->srcpad) ? audioresample->sinkpad :
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audioresample->srcpad;
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structure = gst_caps_get_structure (caps, 0);
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ret = gst_structure_get_int (structure, "rate", &rate);
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ret &= gst_structure_get_int (structure, "channels", &channels);
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if (!ret)
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{
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return GST_PAD_LINK_REFUSED;
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}
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if (gst_pad_is_negotiated (otherpad))
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{
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GstCaps *othercaps = gst_caps_copy (caps);
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int otherrate;
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GstPadLinkReturn linkret;
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if (pad == audioresample->srcpad) {
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otherrate = audioresample->i_rate;
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} else {
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otherrate = audioresample->o_rate;
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}
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gst_caps_set_simple (othercaps, "rate", G_TYPE_INT, otherrate, NULL);
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linkret = gst_pad_try_set_caps (otherpad, othercaps);
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if (GST_PAD_LINK_FAILED (linkret)) {
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return GST_PAD_LINK_REFUSED;
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}
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}
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audioresample->channels = channels;
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resample_set_n_channels (audioresample->resample, audioresample->channels);
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if (pad == audioresample->srcpad) {
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audioresample->o_rate = rate;
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resample_set_output_rate (audioresample->resample, audioresample->o_rate);
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GST_DEBUG ("set o_rate to %d", rate);
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} else {
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audioresample->i_rate = rate;
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resample_set_input_rate (audioresample->resample, audioresample->i_rate);
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GST_DEBUG ("set i_rate to %d", rate);
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}
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return GST_PAD_LINK_OK;
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}
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static void gst_audioresample_init (Audioresample * audioresample)
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static void gst_audioresample_init (GstAudioresample * audioresample)
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{
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ResampleState *r;
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audioresample->sinkpad =
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gst_pad_new_from_template (gst_static_pad_template_get
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(&gst_audioresample_sink_template), "sink");
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gst_element_add_pad (GST_ELEMENT (audioresample), audioresample->sinkpad);
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gst_pad_set_chain_function (audioresample->sinkpad, gst_audioresample_chain);
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gst_pad_set_link_function (audioresample->sinkpad, gst_audioresample_link);
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gst_pad_set_getcaps_function (audioresample->sinkpad,
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gst_audioresample_getcaps);
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gst_pad_set_fixate_function (audioresample->sinkpad,
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gst_audioresample_fixate);
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audioresample->srcpad =
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gst_pad_new_from_template (gst_static_pad_template_get
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(&gst_audioresample_src_template), "src");
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gst_element_add_pad (GST_ELEMENT (audioresample), audioresample->srcpad);
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gst_pad_set_link_function (audioresample->srcpad, gst_audioresample_link);
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gst_pad_set_getcaps_function (audioresample->srcpad,
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gst_audioresample_getcaps);
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gst_pad_set_fixate_function (audioresample->srcpad, gst_audioresample_fixate);
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r = resample_new ();
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audioresample->resample = r;
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@ -311,74 +162,245 @@ static void gst_audioresample_init (Audioresample * audioresample)
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static void gst_audioresample_dispose (GObject * object)
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{
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Audioresample *audioresample = GST_AUDIORESAMPLE (object);
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GstAudioresample *audioresample = GST_AUDIORESAMPLE (object);
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if (audioresample->resample) {
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resample_free (audioresample->resample);
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audioresample->resample = NULL;
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}
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G_OBJECT_CLASS (parent_class)->dispose (object);
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}
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static void gst_audioresample_chain (GstPad * pad, GstData * _data)
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/* vmethods */
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gboolean
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audioresample_get_unit_size (GstBaseTransform * base, GstCaps * caps,
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guint * size) {
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gint width, channels;
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GstStructure *structure;
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gboolean ret;
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g_return_val_if_fail (size, FALSE);
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/* this works for both float and int */
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structure = gst_caps_get_structure (caps, 0);
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ret = gst_structure_get_int (structure, "width", &width);
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ret &= gst_structure_get_int (structure, "channels", &channels);
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g_return_val_if_fail (ret, FALSE);
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*size = width * channels / 8;
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return TRUE;
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}
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GstCaps *audioresample_transform_caps (GstBaseTransform * base,
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GstPadDirection direction, GstCaps * caps)
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{
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GstBuffer *buf = GST_BUFFER (_data);
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Audioresample *audioresample;
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GstCaps *temp, *res;
|
||||
const GstCaps *templcaps;
|
||||
GstStructure *structure;
|
||||
|
||||
temp = gst_caps_copy (caps);
|
||||
structure = gst_caps_get_structure (temp, 0);
|
||||
gst_structure_remove_field (structure, "rate");
|
||||
templcaps = gst_pad_get_pad_template_caps (base->srcpad);
|
||||
res = gst_caps_intersect (templcaps, temp);
|
||||
gst_caps_unref (temp);
|
||||
|
||||
return res;
|
||||
}
|
||||
|
||||
static gboolean
|
||||
resample_set_state_from_caps (ResampleState * state, GstCaps * incaps,
|
||||
GstCaps * outcaps, gint * channels, gint * inrate, gint * outrate)
|
||||
{
|
||||
GstStructure *structure;
|
||||
gboolean ret;
|
||||
gint myinrate, myoutrate;
|
||||
int mychannels;
|
||||
|
||||
GST_DEBUG ("incaps %" GST_PTR_FORMAT ", outcaps %"
|
||||
GST_PTR_FORMAT, incaps, outcaps);
|
||||
|
||||
structure = gst_caps_get_structure (incaps, 0);
|
||||
|
||||
/* FIXME: once it does float, set the correct format */
|
||||
#if 0
|
||||
if (g_str_equal (gst_structure_get_name (structure), "audio/x-raw-float")) {
|
||||
r->format = GST_RESAMPLE_FLOAT;
|
||||
} else {
|
||||
r->format = GST_RESAMPLE_S16;
|
||||
}
|
||||
#endif
|
||||
|
||||
ret = gst_structure_get_int (structure, "rate", &myinrate);
|
||||
ret &= gst_structure_get_int (structure, "channels", &mychannels);
|
||||
g_return_val_if_fail (ret, FALSE);
|
||||
|
||||
structure = gst_caps_get_structure (outcaps, 0);
|
||||
ret = gst_structure_get_int (structure, "rate", &myoutrate);
|
||||
g_return_val_if_fail (ret, FALSE);
|
||||
|
||||
if (channels)
|
||||
*channels = mychannels;
|
||||
if (inrate)
|
||||
*inrate = myinrate;
|
||||
if (outrate)
|
||||
*outrate = myoutrate;
|
||||
|
||||
resample_set_n_channels (state, mychannels);
|
||||
resample_set_input_rate (state, myinrate);
|
||||
resample_set_output_rate (state, myoutrate);
|
||||
|
||||
return TRUE;
|
||||
}
|
||||
|
||||
gboolean audioresample_transform_size (GstBaseTransform * base,
|
||||
GstPadDirection direction, GstCaps * caps, guint size, GstCaps * othercaps,
|
||||
guint * othersize)
|
||||
{
|
||||
GstAudioresample *audioresample = GST_AUDIORESAMPLE (base);
|
||||
ResampleState *state;
|
||||
GstCaps *srccaps, *sinkcaps;
|
||||
gboolean use_internal = FALSE; /* whether we use the internal state */
|
||||
gboolean ret = TRUE;
|
||||
|
||||
/* FIXME: make sure incaps/outcaps get renamed to caps/othercaps, since
|
||||
* interpretation depends on the direction */
|
||||
if (direction == GST_PAD_SINK) {
|
||||
sinkcaps = caps;
|
||||
srccaps = othercaps;
|
||||
} else {
|
||||
sinkcaps = othercaps;
|
||||
srccaps = caps;
|
||||
}
|
||||
|
||||
/* if the caps are the ones that _set_caps got called with; we can use
|
||||
* our own state; otherwise we'll have to create a state */
|
||||
if (gst_caps_is_equal (sinkcaps, audioresample->sinkcaps) &&
|
||||
gst_caps_is_equal (srccaps, audioresample->srccaps)) {
|
||||
use_internal = TRUE;
|
||||
state = audioresample->resample;
|
||||
} else {
|
||||
state = resample_new ();
|
||||
resample_set_state_from_caps (state, sinkcaps, srccaps, NULL, NULL, NULL);
|
||||
}
|
||||
|
||||
/* we can use our own state to answer the question */
|
||||
if (direction == GST_PAD_SINK) {
|
||||
/* asked to convert size of an incoming buffer */
|
||||
*othersize = resample_get_output_size_for_input (state, size);
|
||||
} else {
|
||||
/* take a best guess, this is called cheating */
|
||||
*othersize = floor (size * state->i_rate / state->o_rate);
|
||||
}
|
||||
|
||||
if (!use_internal) {
|
||||
resample_free (state);
|
||||
}
|
||||
|
||||
return ret;
|
||||
}
|
||||
|
||||
gboolean audioresample_set_caps (GstBaseTransform * base, GstCaps * incaps,
|
||||
GstCaps * outcaps)
|
||||
{
|
||||
gboolean ret;
|
||||
gint inrate, outrate;
|
||||
int channels;
|
||||
GstAudioresample *audioresample = GST_AUDIORESAMPLE (base);
|
||||
|
||||
GST_DEBUG_OBJECT (base, "incaps %" GST_PTR_FORMAT ", outcaps %"
|
||||
GST_PTR_FORMAT, incaps, outcaps);
|
||||
|
||||
ret = resample_set_state_from_caps (audioresample->resample, incaps, outcaps,
|
||||
&channels, &inrate, &outrate);
|
||||
|
||||
g_return_val_if_fail (ret, FALSE);
|
||||
|
||||
audioresample->channels = channels;
|
||||
GST_DEBUG_OBJECT (audioresample, "set channels to %d", channels);
|
||||
audioresample->i_rate = inrate;
|
||||
GST_DEBUG_OBJECT (audioresample, "set i_rate to %d", inrate);
|
||||
audioresample->o_rate = outrate;
|
||||
GST_DEBUG_OBJECT (audioresample, "set o_rate to %d", outrate);
|
||||
|
||||
/* save caps so we can short-circuit in the size_transform if the caps
|
||||
* are the same */
|
||||
/* FIXME: clean them up in state change ? */
|
||||
gst_caps_ref (incaps);
|
||||
gst_caps_replace (&audioresample->sinkcaps, incaps);
|
||||
gst_caps_ref (outcaps);
|
||||
gst_caps_replace (&audioresample->srccaps, outcaps);
|
||||
|
||||
return TRUE;
|
||||
}
|
||||
|
||||
static GstFlowReturn
|
||||
audioresample_transform (GstBaseTransform * base, GstBuffer * inbuf,
|
||||
GstBuffer * outbuf)
|
||||
{
|
||||
/* FIXME: this-> */
|
||||
GstAudioresample *audioresample = GST_AUDIORESAMPLE (base);
|
||||
ResampleState *r;
|
||||
guchar *data;
|
||||
gulong size;
|
||||
int outsize;
|
||||
GstBuffer *outbuf;
|
||||
|
||||
g_return_if_fail (pad != NULL);
|
||||
g_return_if_fail (GST_IS_PAD (pad));
|
||||
g_return_if_fail (buf != NULL);
|
||||
|
||||
audioresample = GST_AUDIORESAMPLE (gst_pad_get_parent (pad));
|
||||
|
||||
if (!GST_IS_BUFFER (_data)) {
|
||||
gst_pad_push (audioresample->srcpad, _data);
|
||||
return;
|
||||
}
|
||||
|
||||
/* FIXME: move to _inplace */
|
||||
#if 0
|
||||
if (audioresample->passthru) {
|
||||
gst_pad_push (audioresample->srcpad, GST_DATA (buf));
|
||||
return;
|
||||
}
|
||||
#endif
|
||||
|
||||
r = audioresample->resample;
|
||||
|
||||
data = GST_BUFFER_DATA (buf);
|
||||
size = GST_BUFFER_SIZE (buf);
|
||||
data = GST_BUFFER_DATA (inbuf);
|
||||
size = GST_BUFFER_SIZE (inbuf);
|
||||
|
||||
GST_DEBUG ("got buffer of %ld bytes", size);
|
||||
GST_DEBUG_OBJECT (audioresample, "got buffer of %ld bytes", size);
|
||||
|
||||
resample_add_input_data (r, data, size, (ResampleCallback) gst_data_unref,
|
||||
buf);
|
||||
resample_add_input_data (r, data, size, NULL, NULL);
|
||||
|
||||
outsize = resample_get_output_size (r);
|
||||
/* FIXME this is audioresample being dumb. dunno why */
|
||||
if (outsize == 0) {
|
||||
GST_ERROR ("overriding outbuf size");
|
||||
outsize = size;
|
||||
if (outsize != GST_BUFFER_SIZE (outbuf)) {
|
||||
GST_WARNING_OBJECT (audioresample,
|
||||
"overriding audioresample's outsize %d with outbuffer's size %d",
|
||||
outsize, GST_BUFFER_SIZE (outbuf));
|
||||
outsize = GST_BUFFER_SIZE (outbuf);
|
||||
}
|
||||
outbuf = gst_buffer_new_and_alloc (outsize);
|
||||
|
||||
outsize = resample_get_output_data (r, GST_BUFFER_DATA (outbuf), outsize);
|
||||
GST_BUFFER_SIZE (outbuf) = outsize;
|
||||
|
||||
GST_BUFFER_TIMESTAMP (outbuf) =
|
||||
audioresample->offset * GST_SECOND / audioresample->o_rate;
|
||||
audioresample->offset += outsize / sizeof (gint16) / audioresample->channels;
|
||||
|
||||
gst_pad_push (audioresample->srcpad, GST_DATA (outbuf));
|
||||
if (outsize != GST_BUFFER_SIZE (outbuf)) {
|
||||
GST_WARNING_OBJECT (audioresample,
|
||||
"audioresample, you bastard ! you only gave me %d bytes, not %d",
|
||||
outsize, GST_BUFFER_SIZE (outbuf));
|
||||
/* if the size we get is smaller than the buffer, it's still fine; we
|
||||
* just waste a bit of space on the end */
|
||||
if (outsize < GST_BUFFER_SIZE (outbuf)) {
|
||||
GST_BUFFER_SIZE (outbuf) = outsize;
|
||||
return GST_FLOW_OK;
|
||||
} else {
|
||||
/* this is an error that needs fixing in the resample library; we told
|
||||
* it we wanted only GST_BUFFER_SIZE (outbuf), and it gave us more ! */
|
||||
return GST_FLOW_ERROR;
|
||||
}
|
||||
}
|
||||
|
||||
return GST_FLOW_OK;
|
||||
}
|
||||
|
||||
static void
|
||||
gst_audioresample_set_property (GObject * object, guint prop_id,
|
||||
const GValue * value, GParamSpec * pspec)
|
||||
{
|
||||
Audioresample *audioresample;
|
||||
GstAudioresample *audioresample;
|
||||
|
||||
g_return_if_fail (GST_IS_AUDIORESAMPLE (object));
|
||||
audioresample = GST_AUDIORESAMPLE (object);
|
||||
|
|
@ -400,7 +422,7 @@ static void
|
|||
gst_audioresample_get_property (GObject * object, guint prop_id,
|
||||
GValue * value, GParamSpec * pspec)
|
||||
{
|
||||
Audioresample *audioresample;
|
||||
GstAudioresample *audioresample;
|
||||
|
||||
g_return_if_fail (GST_IS_AUDIORESAMPLE (object));
|
||||
audioresample = GST_AUDIORESAMPLE (object);
|
||||
|
|
@ -431,4 +453,4 @@ static gboolean plugin_init (GstPlugin * plugin)
|
|||
GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
|
||||
GST_VERSION_MINOR,
|
||||
"audioresample",
|
||||
"Resamples audio", plugin_init, VERSION, "LGPL", GST_PACKAGE, GST_ORIGIN)
|
||||
"Resamples audio", plugin_init, VERSION, "LGPL", GST_PACKAGE, GST_ORIGIN);
|
||||
|
|
|
|||
|
|
@ -23,31 +23,32 @@
|
|||
|
||||
|
||||
#include <gst/gst.h>
|
||||
#include <gst/base/gstbasetransform.h>
|
||||
|
||||
#include <audioresample/resample.h>
|
||||
#include "resample.h"
|
||||
|
||||
|
||||
G_BEGIN_DECLS
|
||||
|
||||
|
||||
#define GST_TYPE_AUDIORESAMPLE \
|
||||
(audioresample_get_type())
|
||||
(gst_audioresample_get_type())
|
||||
#define GST_AUDIORESAMPLE(obj) \
|
||||
(G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIORESAMPLE,Audioresample))
|
||||
(G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIORESAMPLE,GstAudioresample))
|
||||
#define GST_AUDIORESAMPLE_CLASS(klass) \
|
||||
(G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_AUDIORESAMPLE,Audioresample))
|
||||
(G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_AUDIORESAMPLE,GstAudioresample))
|
||||
#define GST_IS_AUDIORESAMPLE(obj) \
|
||||
(G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIORESAMPLE))
|
||||
#define GST_IS_AUDIORESAMPLE_CLASS(obj) \
|
||||
(G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_AUDIORESAMPLE))
|
||||
|
||||
typedef struct _Audioresample Audioresample;
|
||||
typedef struct _AudioresampleClass AudioresampleClass;
|
||||
typedef struct _GstAudioresample GstAudioresample;
|
||||
typedef struct _GstAudioresampleClass GstAudioresampleClass;
|
||||
|
||||
struct _Audioresample {
|
||||
GstElement element;
|
||||
struct _GstAudioresample {
|
||||
GstBaseTransform element;
|
||||
|
||||
GstPad *sinkpad,*srcpad;
|
||||
GstCaps *srccaps, *sinkcaps;
|
||||
|
||||
gboolean passthru;
|
||||
|
||||
|
|
@ -61,8 +62,8 @@ struct _Audioresample {
|
|||
ResampleState * resample;
|
||||
};
|
||||
|
||||
struct _AudioresampleClass {
|
||||
GstElementClass parent_class;
|
||||
struct _GstAudioresampleClass {
|
||||
GstBaseTransformClass parent_class;
|
||||
};
|
||||
|
||||
GType gst_audioresample_get_type(void);
|
||||
|
|
|
|||
|
|
@ -29,9 +29,9 @@
|
|||
#include <limits.h>
|
||||
#include <liboil/liboil.h>
|
||||
|
||||
#include <audioresample/resample.h>
|
||||
#include <audioresample/buffer.h>
|
||||
#include <audioresample/debug.h>
|
||||
#include "resample.h"
|
||||
#include "buffer.h"
|
||||
#include "debug.h"
|
||||
|
||||
void resample_scale_ref (ResampleState * r);
|
||||
void resample_scale_functable (ResampleState * r);
|
||||
|
|
@ -101,6 +101,10 @@ resample_buffer_free (AudioresampleBuffer * buffer, void *priv)
|
|||
}
|
||||
}
|
||||
|
||||
/**
|
||||
* free_func: a function that frees the given closure. If NULL, caller is
|
||||
* responsible for freeing.
|
||||
*/
|
||||
void
|
||||
resample_add_input_data (ResampleState * r, void *data, int size,
|
||||
void (*free_func) (void *), void *closure)
|
||||
|
|
@ -134,6 +138,12 @@ resample_input_eos (ResampleState * r)
|
|||
r->eos = 1;
|
||||
}
|
||||
|
||||
int
|
||||
resample_get_output_size_for_input (ResampleState * r, int size)
|
||||
{
|
||||
return floor (size * r->o_rate / r->i_rate);
|
||||
}
|
||||
|
||||
int
|
||||
resample_get_output_size (ResampleState * r)
|
||||
{
|
||||
|
|
|
|||
|
|
@ -21,8 +21,8 @@
|
|||
#ifndef __RESAMPLE_H__
|
||||
#define __RESAMPLE_H__
|
||||
|
||||
#include <audioresample/functable.h>
|
||||
#include <audioresample/buffer.h>
|
||||
#include "functable.h"
|
||||
#include "buffer.h"
|
||||
|
||||
typedef enum {
|
||||
RESAMPLE_FORMAT_S16 = 0,
|
||||
|
|
@ -89,8 +89,8 @@ struct _ResampleState {
|
|||
double *out_tmp;
|
||||
};
|
||||
|
||||
void resample_init(void);
|
||||
void resample_cleanup(void);
|
||||
void resample_init (void);
|
||||
void resample_cleanup (void);
|
||||
|
||||
ResampleState *resample_new (void);
|
||||
void resample_free (ResampleState *state);
|
||||
|
|
@ -98,6 +98,8 @@ void resample_free (ResampleState *state);
|
|||
void resample_add_input_data (ResampleState * r, void *data, int size,
|
||||
ResampleCallback free_func, void *closure);
|
||||
void resample_input_eos (ResampleState *r);
|
||||
|
||||
int resample_get_output_size_for_input (ResampleState * r, int size);
|
||||
int resample_get_output_size (ResampleState *r);
|
||||
int resample_get_output_data (ResampleState *r, void *data, int size);
|
||||
|
||||
|
|
@ -109,6 +111,5 @@ void resample_set_format (ResampleState *r, ResampleFormat format);
|
|||
void resample_set_method (ResampleState *r, int method);
|
||||
int resample_format_size (ResampleFormat format);
|
||||
|
||||
|
||||
#endif /* __RESAMPLE_H__ */
|
||||
|
||||
|
|
|
|||
|
|
@ -29,9 +29,9 @@
|
|||
#include <limits.h>
|
||||
#include <liboil/liboil.h>
|
||||
|
||||
#include <audioresample/resample.h>
|
||||
#include <audioresample/buffer.h>
|
||||
#include <audioresample/debug.h>
|
||||
#include "resample.h"
|
||||
#include "buffer.h"
|
||||
#include "debug.h"
|
||||
|
||||
|
||||
static double
|
||||
|
|
|
|||
|
|
@ -29,9 +29,9 @@
|
|||
#include <limits.h>
|
||||
#include <liboil/liboil.h>
|
||||
|
||||
#include <audioresample/resample.h>
|
||||
#include <audioresample/buffer.h>
|
||||
#include <audioresample/debug.h>
|
||||
#include "resample.h"
|
||||
#include "buffer.h"
|
||||
#include "debug.h"
|
||||
|
||||
static void
|
||||
func_sinc (double *fx, double *dfx, double x, void *closure)
|
||||
|
|
|
|||
|
|
@ -29,9 +29,9 @@
|
|||
#include <limits.h>
|
||||
#include <liboil/liboil.h>
|
||||
|
||||
#include <audioresample/resample.h>
|
||||
#include <audioresample/buffer.h>
|
||||
#include <audioresample/debug.h>
|
||||
#include "resample.h"
|
||||
#include "buffer.h"
|
||||
#include "debug.h"
|
||||
|
||||
|
||||
static double
|
||||
|
|
|
|||
Loading…
Reference in a new issue