diff options
Diffstat (limited to 'gst')
| -rw-r--r-- | gst/audioresample/Makefile.am | 4 | ||||
| -rw-r--r-- | gst/audioresample/buffer.c | 5 | ||||
| -rw-r--r-- | gst/audioresample/functable.c | 4 | ||||
| -rw-r--r-- | gst/audioresample/gstaudioresample.c | 426 | ||||
| -rw-r--r-- | gst/audioresample/gstaudioresample.h | 23 | ||||
| -rw-r--r-- | gst/audioresample/resample.c | 16 | ||||
| -rw-r--r-- | gst/audioresample/resample.h | 11 | ||||
| -rw-r--r-- | gst/audioresample/resample_chunk.c | 6 | ||||
| -rw-r--r-- | gst/audioresample/resample_functable.c | 6 | ||||
| -rw-r--r-- | gst/audioresample/resample_ref.c | 6 |
10 files changed, 271 insertions, 236 deletions
diff --git a/gst/audioresample/Makefile.am b/gst/audioresample/Makefile.am index bff05034..36fd56c5 100644 --- a/gst/audioresample/Makefile.am +++ b/gst/audioresample/Makefile.am @@ -15,7 +15,7 @@ resample_SOURCES = \ buffer.h libgstaudioresample_la_SOURCES = gstaudioresample.c $(resample_SOURCES) -libgstaudioresample_la_CFLAGS = $(GST_CFLAGS) $(LIBOIL_CFLAGS) -libgstaudioresample_la_LIBADD = $(LIBOIL_LIBS) +libgstaudioresample_la_CFLAGS = $(GST_BASE_CFLAGS) $(GST_CFLAGS) $(LIBOIL_CFLAGS) +libgstaudioresample_la_LIBADD = $(GST_BASE_LIBS) $(GST_LIBS) $(LIBOIL_LIBS) libgstaudioresample_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS) diff --git a/gst/audioresample/buffer.c b/gst/audioresample/buffer.c index f72e6056..679fa020 100644 --- a/gst/audioresample/buffer.c +++ b/gst/audioresample/buffer.c @@ -3,10 +3,11 @@ #include "config.h" #endif -#include <audioresample/buffer.h> #include <glib.h> #include <string.h> -#include <audioresample/debug.h> + +#include "buffer.h" +#include "debug.h" static void audioresample_buffer_free_mem (AudioresampleBuffer * buffer, void *); diff --git a/gst/audioresample/functable.c b/gst/audioresample/functable.c index 41844015..d627361f 100644 --- a/gst/audioresample/functable.c +++ b/gst/audioresample/functable.c @@ -26,8 +26,8 @@ #include <stdio.h> #include <stdlib.h> -#include <audioresample/functable.h> -#include <audioresample/debug.h> +#include "functable.h" +#include "debug.h" diff --git a/gst/audioresample/gstaudioresample.c b/gst/audioresample/gstaudioresample.c index 363acd9b..f2549b28 100644 --- a/gst/audioresample/gstaudioresample.c +++ b/gst/audioresample/gstaudioresample.c @@ -1,5 +1,5 @@ /* GStreamer - * Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu> + * Copyright (C) 1999 Erik Walthinsen <omega@cse.ogi.edu> * Copyright (C) 2003,2004 David A. Schleef <ds@schleef.org> * * This library is free software; you can redistribute it and/or @@ -19,16 +19,17 @@ */ /* Element-Checklist-Version: 5 */ - #ifdef HAVE_CONFIG_H #include "config.h" #endif + #include <string.h> #include <math.h> /*#define DEBUG_ENABLED */ #include "gstaudioresample.h" #include <gst/audio/audio.h> +#include <gst/base/gstbasetransform.h> GST_DEBUG_CATEGORY_STATIC (audioresample_debug); #define GST_CAT_DEFAULT audioresample_debug @@ -40,7 +41,7 @@ GST_ELEMENT_DETAILS ("Audio scaler", "Resample audio", "David Schleef <ds@schleef.org>"); -/* Audioresample signals and args */ +/* GstAudioresample signals and args */ enum { /* FILL ME */ @@ -79,63 +80,54 @@ enum GST_PAD_SRC, GST_PAD_ALWAYS, SUPPORTED_CAPS); static void gst_audioresample_base_init (gpointer g_class); - static void gst_audioresample_class_init (AudioresampleClass * klass); - static void gst_audioresample_init (Audioresample * audioresample); + static void gst_audioresample_class_init (GstAudioresampleClass * klass); + static void gst_audioresample_init (GstAudioresample * audioresample); static void gst_audioresample_dispose (GObject * object); - static void gst_audioresample_chain (GstPad * pad, GstData * _data); - static void gst_audioresample_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec); static void gst_audioresample_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec); - static GstElementClass *parent_class = NULL; +/* vmethods */ + gboolean audioresample_get_unit_size (GstBaseTransform * base, + GstCaps * caps, guint * size); + GstCaps *audioresample_transform_caps (GstBaseTransform * base, + GstPadDirection direction, GstCaps * caps); + gboolean audioresample_transform_size (GstBaseTransform * trans, + GstPadDirection direction, GstCaps * incaps, guint insize, + GstCaps * outcaps, guint * outsize); + gboolean audioresample_set_caps (GstBaseTransform * base, GstCaps * incaps, + GstCaps * outcaps); + static GstFlowReturn audioresample_transform (GstBaseTransform * base, + GstBuffer * inbuf, GstBuffer * outbuf); /*static guint gst_audioresample_signals[LAST_SIGNAL] = { 0 }; */ - GType audioresample_get_type (void) - { - static GType audioresample_type = 0; - - if (!audioresample_type) - { - static const GTypeInfo audioresample_info = { - sizeof (AudioresampleClass), - gst_audioresample_base_init, - NULL, - (GClassInitFunc) gst_audioresample_class_init, - NULL, - NULL, - sizeof (Audioresample), 0, - (GInstanceInitFunc) gst_audioresample_init,}; - - audioresample_type = - g_type_register_static (GST_TYPE_ELEMENT, "Audioresample", - &audioresample_info, 0); - } - return audioresample_type; - } +#define DEBUG_INIT(bla) \ + GST_DEBUG_CATEGORY_INIT (audioresample_debug, "audioresample", 0, "audio resampling element"); -static void gst_audioresample_base_init (gpointer g_class) -{ - GstElementClass *gstelement_class = GST_ELEMENT_CLASS (g_class); +GST_BOILERPLATE_FULL (GstAudioresample, gst_audioresample, GstBaseTransform, + GST_TYPE_BASE_TRANSFORM, DEBUG_INIT); - gst_element_class_add_pad_template (gstelement_class, - gst_static_pad_template_get (&gst_audioresample_src_template)); - gst_element_class_add_pad_template (gstelement_class, - gst_static_pad_template_get (&gst_audioresample_sink_template)); + static void gst_audioresample_base_init (gpointer g_class) + { + GstElementClass *gstelement_class = GST_ELEMENT_CLASS (g_class); - gst_element_class_set_details (gstelement_class, &gst_audioresample_details); -} + gst_element_class_add_pad_template (gstelement_class, + gst_static_pad_template_get (&gst_audioresample_src_template)); + gst_element_class_add_pad_template (gstelement_class, + gst_static_pad_template_get (&gst_audioresample_sink_template)); -static void gst_audioresample_class_init (AudioresampleClass * klass) + gst_element_class_set_details (gstelement_class, + &gst_audioresample_details); + } + +static void gst_audioresample_class_init (GstAudioresampleClass * klass) { GObjectClass *gobject_class; - GstElementClass *gstelement_class; gobject_class = (GObjectClass *) klass; - gstelement_class = (GstElementClass *) klass; gobject_class->set_property = gst_audioresample_set_property; gobject_class->get_property = gst_audioresample_get_property; @@ -145,240 +137,270 @@ static void gst_audioresample_class_init (AudioresampleClass * klass) g_param_spec_int ("filter_length", "filter_length", "filter_length", 0, G_MAXINT, 16, G_PARAM_READWRITE | G_PARAM_CONSTRUCT)); - parent_class = g_type_class_ref (GST_TYPE_ELEMENT); - - GST_DEBUG_CATEGORY_INIT (audioresample_debug, "audioresample", 0, - "audioresample element"); + GST_BASE_TRANSFORM_CLASS (klass)->transform_size = + GST_DEBUG_FUNCPTR (audioresample_transform_size); + GST_BASE_TRANSFORM_CLASS (klass)->get_unit_size = + GST_DEBUG_FUNCPTR (audioresample_get_unit_size); + GST_BASE_TRANSFORM_CLASS (klass)->transform_caps = + GST_DEBUG_FUNCPTR (audioresample_transform_caps); + GST_BASE_TRANSFORM_CLASS (klass)->set_caps = + GST_DEBUG_FUNCPTR (audioresample_set_caps); + GST_BASE_TRANSFORM_CLASS (klass)->transform = + GST_DEBUG_FUNCPTR (audioresample_transform); } -static void gst_audioresample_expand_caps (GstCaps * caps) +static void gst_audioresample_init (GstAudioresample * audioresample) { - gint i; + ResampleState *r; - for (i = 0; i < gst_caps_get_size (caps); i++) { - GstStructure *structure = gst_caps_get_structure (caps, i); - const GValue *value; + r = resample_new (); + audioresample->resample = r; - value = gst_structure_get_value (structure, "rate"); - if (value == NULL) { - GST_ERROR ("caps structure doesn't have required rate field"); - return; - } + resample_set_filter_length (r, 64); + resample_set_format (r, RESAMPLE_FORMAT_S16); +} + +static void gst_audioresample_dispose (GObject * object) +{ + GstAudioresample *audioresample = GST_AUDIORESAMPLE (object); - gst_structure_set (structure, "rate", GST_TYPE_INT_RANGE, 1, G_MAXINT, 0); + if (audioresample->resample) { + resample_free (audioresample->resample); + audioresample->resample = NULL; } + + G_OBJECT_CLASS (parent_class)->dispose (object); } -static GstCaps *gst_audioresample_getcaps (GstPad * pad) -{ - Audioresample *audioresample; - GstCaps *caps; - GstPad *otherpad; +/* vmethods */ +gboolean + audioresample_get_unit_size (GstBaseTransform * base, GstCaps * caps, + guint * size) { + gint width, channels; + GstStructure *structure; + gboolean ret; - audioresample = GST_AUDIORESAMPLE (gst_pad_get_parent (pad)); + g_return_val_if_fail (size, FALSE); - otherpad = (pad == audioresample->srcpad) ? audioresample->sinkpad : - audioresample->srcpad; - caps = gst_pad_get_allowed_caps (otherpad); + /* this works for both float and int */ + structure = gst_caps_get_structure (caps, 0); + ret = gst_structure_get_int (structure, "width", &width); + ret &= gst_structure_get_int (structure, "channels", &channels); + g_return_val_if_fail (ret, FALSE); - gst_audioresample_expand_caps (caps); + *size = width * channels / 8; - return caps; + return TRUE; } -static GstCaps *gst_audioresample_fixate (GstPad * pad, const GstCaps * caps) +GstCaps *audioresample_transform_caps (GstBaseTransform * base, + GstPadDirection direction, GstCaps * caps) { - Audioresample *audioresample; - GstPad *otherpad; - int rate; - GstCaps *copy; + GstCaps *temp, *res; + const GstCaps *templcaps; GstStructure *structure; - audioresample = GST_AUDIORESAMPLE (gst_pad_get_parent (pad)); + 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); - if (pad == audioresample->srcpad) { - otherpad = audioresample->sinkpad; - rate = audioresample->i_rate; - } else - { - otherpad = audioresample->srcpad; - rate = audioresample->o_rate; - } - if (!GST_PAD_IS_NEGOTIATING (otherpad)) - return NULL; - if (gst_caps_get_size (caps) > 1) - return NULL; - - copy = gst_caps_copy (caps); - structure = gst_caps_get_structure (copy, 0); - if (rate) { - if (gst_caps_structure_fixate_field_nearest_int (structure, "rate", rate)) { - return copy; - } - } - gst_caps_free (copy); - return NULL; + return res; } -static GstPadLinkReturn gst_audioresample_link (GstPad * pad, - const GstCaps * caps) +static gboolean + resample_set_state_from_caps (ResampleState * state, GstCaps * incaps, + GstCaps * outcaps, gint * channels, gint * inrate, gint * outrate) { - Audioresample *audioresample; GstStructure *structure; - int rate; - int channels; gboolean ret; - GstPad *otherpad; + gint myinrate, myoutrate; + int mychannels; - audioresample = GST_AUDIORESAMPLE (gst_pad_get_parent (pad)); + GST_DEBUG ("incaps %" GST_PTR_FORMAT ", outcaps %" + GST_PTR_FORMAT, incaps, outcaps); - otherpad = (pad == audioresample->srcpad) ? audioresample->sinkpad : - audioresample->srcpad; + structure = gst_caps_get_structure (incaps, 0); - structure = gst_caps_get_structure (caps, 0); - ret = gst_structure_get_int (structure, "rate", &rate); - ret &= gst_structure_get_int (structure, "channels", &channels); - if (!ret) - { - return GST_PAD_LINK_REFUSED; + /* 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 - if (gst_pad_is_negotiated (otherpad)) - { - GstCaps *othercaps = gst_caps_copy (caps); - int otherrate; - GstPadLinkReturn linkret; + ret = gst_structure_get_int (structure, "rate", &myinrate); + ret &= gst_structure_get_int (structure, "channels", &mychannels); + g_return_val_if_fail (ret, FALSE); - if (pad == audioresample->srcpad) { - otherrate = audioresample->i_rate; - } else { - otherrate = audioresample->o_rate; - } - gst_caps_set_simple (othercaps, "rate", G_TYPE_INT, otherrate, NULL); - linkret = gst_pad_try_set_caps (otherpad, othercaps); - if (GST_PAD_LINK_FAILED (linkret)) { - return GST_PAD_LINK_REFUSED; - } + 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; - audioresample->channels = channels; - resample_set_n_channels (audioresample->resample, audioresample->channels); - if (pad == audioresample->srcpad) { - audioresample->o_rate = rate; - resample_set_output_rate (audioresample->resample, audioresample->o_rate); - GST_DEBUG ("set o_rate to %d", rate); - } else { - audioresample->i_rate = rate; - resample_set_input_rate (audioresample->resample, audioresample->i_rate); - GST_DEBUG ("set i_rate to %d", rate); - } + resample_set_n_channels (state, mychannels); + resample_set_input_rate (state, myinrate); + resample_set_output_rate (state, myoutrate); - return GST_PAD_LINK_OK; + return TRUE; } -static void gst_audioresample_init (Audioresample * audioresample) +gboolean audioresample_transform_size (GstBaseTransform * base, + GstPadDirection direction, GstCaps * caps, guint size, GstCaps * othercaps, + guint * othersize) { - ResampleState *r; + 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; + } - audioresample->sinkpad = - gst_pad_new_from_template (gst_static_pad_template_get - (&gst_audioresample_sink_template), "sink"); - gst_element_add_pad (GST_ELEMENT (audioresample), audioresample->sinkpad); - gst_pad_set_chain_function (audioresample->sinkpad, gst_audioresample_chain); - gst_pad_set_link_function (audioresample->sinkpad, gst_audioresample_link); - gst_pad_set_getcaps_function (audioresample->sinkpad, - gst_audioresample_getcaps); - gst_pad_set_fixate_function (audioresample->sinkpad, - gst_audioresample_fixate); - - audioresample->srcpad = - gst_pad_new_from_template (gst_static_pad_template_get - (&gst_audioresample_src_template), "src"); - - gst_element_add_pad (GST_ELEMENT (audioresample), audioresample->srcpad); - gst_pad_set_link_function (audioresample->srcpad, gst_audioresample_link); - gst_pad_set_getcaps_function (audioresample->srcpad, - gst_audioresample_getcaps); - gst_pad_set_fixate_function (audioresample->srcpad, gst_audioresample_fixate); + /* 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); + } - r = resample_new (); - audioresample->resample = r; + /* 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); + } - resample_set_filter_length (r, 64); - resample_set_format (r, RESAMPLE_FORMAT_S16); + if (!use_internal) { + resample_free (state); + } + + return ret; } -static void gst_audioresample_dispose (GObject * object) +gboolean audioresample_set_caps (GstBaseTransform * base, GstCaps * incaps, + GstCaps * outcaps) { - Audioresample *audioresample = GST_AUDIORESAMPLE (object); + gboolean ret; + gint inrate, outrate; + int channels; + GstAudioresample *audioresample = GST_AUDIORESAMPLE (base); - if (audioresample->resample) { - resample_free (audioresample->resample); - } + GST_DEBUG_OBJECT (base, "incaps %" GST_PTR_FORMAT ", outcaps %" + GST_PTR_FORMAT, incaps, outcaps); - G_OBJECT_CLASS (parent_class)->dispose (object); + 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 void gst_audioresample_chain (GstPad * pad, GstData * _data) +static GstFlowReturn + audioresample_transform (GstBaseTransform * base, GstBuffer * inbuf, + GstBuffer * outbuf) { - GstBuffer *buf = GST_BUFFER (_data); - Audioresample *audioresample; + /* 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); diff --git a/gst/audioresample/gstaudioresample.h b/gst/audioresample/gstaudioresample.h index fc5115da..99d937bb 100644 --- a/gst/audioresample/gstaudioresample.h +++ b/gst/audioresample/gstaudioresample.h @@ -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); diff --git a/gst/audioresample/resample.c b/gst/audioresample/resample.c index 38c6ba84..e8ec45fb 100644 --- a/gst/audioresample/resample.c +++ b/gst/audioresample/resample.c @@ -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) @@ -135,6 +139,12 @@ resample_input_eos (ResampleState * r) } 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) { return floor (audioresample_buffer_queue_get_depth (r->queue) * r->o_rate / diff --git a/gst/audioresample/resample.h b/gst/audioresample/resample.h index 9be54f46..ea4aa305 100644 --- a/gst/audioresample/resample.h +++ b/gst/audioresample/resample.h @@ -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__ */ diff --git a/gst/audioresample/resample_chunk.c b/gst/audioresample/resample_chunk.c index 53755e62..c91e1f2a 100644 --- a/gst/audioresample/resample_chunk.c +++ b/gst/audioresample/resample_chunk.c @@ -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 diff --git a/gst/audioresample/resample_functable.c b/gst/audioresample/resample_functable.c index af5f9253..7db06ff8 100644 --- a/gst/audioresample/resample_functable.c +++ b/gst/audioresample/resample_functable.c @@ -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) diff --git a/gst/audioresample/resample_ref.c b/gst/audioresample/resample_ref.c index a4623e71..6717aa23 100644 --- a/gst/audioresample/resample_ref.c +++ b/gst/audioresample/resample_ref.c @@ -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 |