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-rw-r--r--gst/legacyresample/Makefile.am (renamed from gst/audioresample/Makefile.am)4
-rw-r--r--gst/legacyresample/buffer.c (renamed from gst/audioresample/buffer.c)0
-rw-r--r--gst/legacyresample/buffer.h (renamed from gst/audioresample/buffer.h)0
-rw-r--r--gst/legacyresample/debug.c (renamed from gst/audioresample/debug.c)0
-rw-r--r--gst/legacyresample/debug.h (renamed from gst/audioresample/debug.h)0
-rw-r--r--gst/legacyresample/functable.c (renamed from gst/audioresample/functable.c)0
-rw-r--r--gst/legacyresample/functable.h (renamed from gst/audioresample/functable.h)0
-rw-r--r--gst/legacyresample/gstlegacyresample.c (renamed from gst/audioresample/gstaudioresample.c)357
-rw-r--r--gst/legacyresample/gstlegacyresample.h (renamed from gst/audioresample/gstaudioresample.h)38
-rw-r--r--gst/legacyresample/resample.c (renamed from gst/audioresample/resample.c)0
-rw-r--r--gst/legacyresample/resample.h (renamed from gst/audioresample/resample.h)0
-rw-r--r--gst/legacyresample/resample_chunk.c (renamed from gst/audioresample/resample_chunk.c)0
-rw-r--r--gst/legacyresample/resample_functable.c (renamed from gst/audioresample/resample_functable.c)0
-rw-r--r--gst/legacyresample/resample_ref.c (renamed from gst/audioresample/resample_ref.c)0
14 files changed, 201 insertions, 198 deletions
diff --git a/gst/audioresample/Makefile.am b/gst/legacyresample/Makefile.am
index c08ab262..41abdc67 100644
--- a/gst/audioresample/Makefile.am
+++ b/gst/legacyresample/Makefile.am
@@ -10,12 +10,12 @@ resample_SOURCES = \
buffer.c
noinst_HEADERS = \
- gstaudioresample.h \
+ gstlegacyresample.h \
functable.h \
debug.h \
buffer.h
-libgstlegacyresample_la_SOURCES = gstaudioresample.c $(resample_SOURCES)
+libgstlegacyresample_la_SOURCES = gstlegacyresample.c $(resample_SOURCES)
libgstlegacyresample_la_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(GST_BASE_CFLAGS) $(GST_CFLAGS) $(LIBOIL_CFLAGS)
libgstlegacyresample_la_LIBADD = $(GST_BASE_LIBS) $(GST_LIBS) $(LIBOIL_LIBS)
libgstlegacyresample_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
diff --git a/gst/audioresample/buffer.c b/gst/legacyresample/buffer.c
index 442b4f8c..442b4f8c 100644
--- a/gst/audioresample/buffer.c
+++ b/gst/legacyresample/buffer.c
diff --git a/gst/audioresample/buffer.h b/gst/legacyresample/buffer.h
index 4cf1fd94..4cf1fd94 100644
--- a/gst/audioresample/buffer.h
+++ b/gst/legacyresample/buffer.h
diff --git a/gst/audioresample/debug.c b/gst/legacyresample/debug.c
index 27877277..27877277 100644
--- a/gst/audioresample/debug.c
+++ b/gst/legacyresample/debug.c
diff --git a/gst/audioresample/debug.h b/gst/legacyresample/debug.h
index ff7deafb..ff7deafb 100644
--- a/gst/audioresample/debug.h
+++ b/gst/legacyresample/debug.h
diff --git a/gst/audioresample/functable.c b/gst/legacyresample/functable.c
index d627361f..d627361f 100644
--- a/gst/audioresample/functable.c
+++ b/gst/legacyresample/functable.c
diff --git a/gst/audioresample/functable.h b/gst/legacyresample/functable.h
index 5f56e2bd..5f56e2bd 100644
--- a/gst/audioresample/functable.h
+++ b/gst/legacyresample/functable.h
diff --git a/gst/audioresample/gstaudioresample.c b/gst/legacyresample/gstlegacyresample.c
index 4f6f85e0..908b6ad9 100644
--- a/gst/audioresample/gstaudioresample.c
+++ b/gst/legacyresample/gstlegacyresample.c
@@ -44,15 +44,15 @@
#include <math.h>
/*#define DEBUG_ENABLED */
-#include "gstaudioresample.h"
+#include "gstlegacyresample.h"
#include <gst/audio/audio.h>
#include <gst/base/gstbasetransform.h>
-GST_DEBUG_CATEGORY_STATIC (audioresample_debug);
-#define GST_CAT_DEFAULT audioresample_debug
+GST_DEBUG_CATEGORY_STATIC (legacyresample_debug);
+#define GST_CAT_DEFAULT legacyresample_debug
/* elementfactory information */
-static const GstElementDetails gst_audioresample_details =
+static const GstElementDetails gst_legacyresample_details =
GST_ELEMENT_DETAILS ("Audio scaler",
"Filter/Converter/Audio",
"Resample audio",
@@ -94,70 +94,71 @@ GST_STATIC_CAPS ( \
"width = (int) 64" \
)
-static GstStaticPadTemplate gst_audioresample_sink_template =
+static GstStaticPadTemplate gst_legacyresample_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK, GST_PAD_ALWAYS, SUPPORTED_CAPS);
-static GstStaticPadTemplate gst_audioresample_src_template =
+static GstStaticPadTemplate gst_legacyresample_src_template =
GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC, GST_PAD_ALWAYS, SUPPORTED_CAPS);
-static void gst_audioresample_set_property (GObject * object,
+static void gst_legacyresample_set_property (GObject * object,
guint prop_id, const GValue * value, GParamSpec * pspec);
-static void gst_audioresample_get_property (GObject * object,
+static void gst_legacyresample_get_property (GObject * object,
guint prop_id, GValue * value, GParamSpec * pspec);
/* vmethods */
-static gboolean audioresample_get_unit_size (GstBaseTransform * base,
+static gboolean legacyresample_get_unit_size (GstBaseTransform * base,
GstCaps * caps, guint * size);
-static GstCaps *audioresample_transform_caps (GstBaseTransform * base,
+static GstCaps *legacyresample_transform_caps (GstBaseTransform * base,
GstPadDirection direction, GstCaps * caps);
-static void audioresample_fixate_caps (GstBaseTransform * base,
+static void legacyresample_fixate_caps (GstBaseTransform * base,
GstPadDirection direction, GstCaps * caps, GstCaps * othercaps);
-static gboolean audioresample_transform_size (GstBaseTransform * trans,
+static gboolean legacyresample_transform_size (GstBaseTransform * trans,
GstPadDirection direction, GstCaps * incaps, guint insize,
GstCaps * outcaps, guint * outsize);
-static gboolean audioresample_set_caps (GstBaseTransform * base,
+static gboolean legacyresample_set_caps (GstBaseTransform * base,
GstCaps * incaps, GstCaps * outcaps);
-static GstFlowReturn audioresample_pushthrough (GstAudioresample *
- audioresample);
-static GstFlowReturn audioresample_transform (GstBaseTransform * base,
+static GstFlowReturn legacyresample_pushthrough (GstLegacyresample *
+ legacyresample);
+static GstFlowReturn legacyresample_transform (GstBaseTransform * base,
GstBuffer * inbuf, GstBuffer * outbuf);
-static gboolean audioresample_event (GstBaseTransform * base, GstEvent * event);
-static gboolean audioresample_start (GstBaseTransform * base);
-static gboolean audioresample_stop (GstBaseTransform * base);
+static gboolean legacyresample_event (GstBaseTransform * base,
+ GstEvent * event);
+static gboolean legacyresample_start (GstBaseTransform * base);
+static gboolean legacyresample_stop (GstBaseTransform * base);
-static gboolean audioresample_query (GstPad * pad, GstQuery * query);
-static const GstQueryType *audioresample_query_type (GstPad * pad);
+static gboolean legacyresample_query (GstPad * pad, GstQuery * query);
+static const GstQueryType *legacyresample_query_type (GstPad * pad);
#define DEBUG_INIT(bla) \
- GST_DEBUG_CATEGORY_INIT (audioresample_debug, "legacyresample", 0, "audio resampling element");
+ GST_DEBUG_CATEGORY_INIT (legacyresample_debug, "legacyresample", 0, "audio resampling element");
-GST_BOILERPLATE_FULL (GstAudioresample, gst_audioresample, GstBaseTransform,
+GST_BOILERPLATE_FULL (GstLegacyresample, gst_legacyresample, GstBaseTransform,
GST_TYPE_BASE_TRANSFORM, DEBUG_INIT);
static void
-gst_audioresample_base_init (gpointer g_class)
+gst_legacyresample_base_init (gpointer g_class)
{
GstElementClass *gstelement_class = GST_ELEMENT_CLASS (g_class);
gst_element_class_add_pad_template (gstelement_class,
- gst_static_pad_template_get (&gst_audioresample_src_template));
+ gst_static_pad_template_get (&gst_legacyresample_src_template));
gst_element_class_add_pad_template (gstelement_class,
- gst_static_pad_template_get (&gst_audioresample_sink_template));
+ gst_static_pad_template_get (&gst_legacyresample_sink_template));
- gst_element_class_set_details (gstelement_class, &gst_audioresample_details);
+ gst_element_class_set_details (gstelement_class, &gst_legacyresample_details);
}
static void
-gst_audioresample_class_init (GstAudioresampleClass * klass)
+gst_legacyresample_class_init (GstLegacyresampleClass * klass)
{
GObjectClass *gobject_class;
gobject_class = (GObjectClass *) klass;
- gobject_class->set_property = gst_audioresample_set_property;
- gobject_class->get_property = gst_audioresample_get_property;
+ gobject_class->set_property = gst_legacyresample_set_property;
+ gobject_class->get_property = gst_legacyresample_get_property;
g_object_class_install_property (gobject_class, PROP_FILTERLEN,
g_param_spec_int ("filter-length", "filter length",
@@ -165,82 +166,82 @@ gst_audioresample_class_init (GstAudioresampleClass * klass)
G_PARAM_READWRITE | G_PARAM_CONSTRUCT | G_PARAM_STATIC_STRINGS));
GST_BASE_TRANSFORM_CLASS (klass)->start =
- GST_DEBUG_FUNCPTR (audioresample_start);
+ GST_DEBUG_FUNCPTR (legacyresample_start);
GST_BASE_TRANSFORM_CLASS (klass)->stop =
- GST_DEBUG_FUNCPTR (audioresample_stop);
+ GST_DEBUG_FUNCPTR (legacyresample_stop);
GST_BASE_TRANSFORM_CLASS (klass)->transform_size =
- GST_DEBUG_FUNCPTR (audioresample_transform_size);
+ GST_DEBUG_FUNCPTR (legacyresample_transform_size);
GST_BASE_TRANSFORM_CLASS (klass)->get_unit_size =
- GST_DEBUG_FUNCPTR (audioresample_get_unit_size);
+ GST_DEBUG_FUNCPTR (legacyresample_get_unit_size);
GST_BASE_TRANSFORM_CLASS (klass)->transform_caps =
- GST_DEBUG_FUNCPTR (audioresample_transform_caps);
+ GST_DEBUG_FUNCPTR (legacyresample_transform_caps);
GST_BASE_TRANSFORM_CLASS (klass)->fixate_caps =
- GST_DEBUG_FUNCPTR (audioresample_fixate_caps);
+ GST_DEBUG_FUNCPTR (legacyresample_fixate_caps);
GST_BASE_TRANSFORM_CLASS (klass)->set_caps =
- GST_DEBUG_FUNCPTR (audioresample_set_caps);
+ GST_DEBUG_FUNCPTR (legacyresample_set_caps);
GST_BASE_TRANSFORM_CLASS (klass)->transform =
- GST_DEBUG_FUNCPTR (audioresample_transform);
+ GST_DEBUG_FUNCPTR (legacyresample_transform);
GST_BASE_TRANSFORM_CLASS (klass)->event =
- GST_DEBUG_FUNCPTR (audioresample_event);
+ GST_DEBUG_FUNCPTR (legacyresample_event);
GST_BASE_TRANSFORM_CLASS (klass)->passthrough_on_same_caps = TRUE;
}
static void
-gst_audioresample_init (GstAudioresample * audioresample,
- GstAudioresampleClass * klass)
+gst_legacyresample_init (GstLegacyresample * legacyresample,
+ GstLegacyresampleClass * klass)
{
GstBaseTransform *trans;
- trans = GST_BASE_TRANSFORM (audioresample);
+ trans = GST_BASE_TRANSFORM (legacyresample);
/* buffer alloc passthrough is too impossible. FIXME, it
* is trivial in the passthrough case. */
gst_pad_set_bufferalloc_function (trans->sinkpad, NULL);
- audioresample->filter_length = DEFAULT_FILTERLEN;
+ legacyresample->filter_length = DEFAULT_FILTERLEN;
- audioresample->need_discont = FALSE;
+ legacyresample->need_discont = FALSE;
- gst_pad_set_query_function (trans->srcpad, audioresample_query);
- gst_pad_set_query_type_function (trans->srcpad, audioresample_query_type);
+ gst_pad_set_query_function (trans->srcpad, legacyresample_query);
+ gst_pad_set_query_type_function (trans->srcpad, legacyresample_query_type);
}
/* vmethods */
static gboolean
-audioresample_start (GstBaseTransform * base)
+legacyresample_start (GstBaseTransform * base)
{
- GstAudioresample *audioresample = GST_AUDIORESAMPLE (base);
+ GstLegacyresample *legacyresample = GST_LEGACYRESAMPLE (base);
- audioresample->resample = resample_new ();
- audioresample->ts_offset = -1;
- audioresample->offset = -1;
- audioresample->next_ts = -1;
+ legacyresample->resample = resample_new ();
+ legacyresample->ts_offset = -1;
+ legacyresample->offset = -1;
+ legacyresample->next_ts = -1;
- resample_set_filter_length (audioresample->resample,
- audioresample->filter_length);
+ resample_set_filter_length (legacyresample->resample,
+ legacyresample->filter_length);
return TRUE;
}
static gboolean
-audioresample_stop (GstBaseTransform * base)
+legacyresample_stop (GstBaseTransform * base)
{
- GstAudioresample *audioresample = GST_AUDIORESAMPLE (base);
+ GstLegacyresample *legacyresample = GST_LEGACYRESAMPLE (base);
- if (audioresample->resample) {
- resample_free (audioresample->resample);
- audioresample->resample = NULL;
+ if (legacyresample->resample) {
+ resample_free (legacyresample->resample);
+ legacyresample->resample = NULL;
}
- gst_caps_replace (&audioresample->sinkcaps, NULL);
- gst_caps_replace (&audioresample->srccaps, NULL);
+ gst_caps_replace (&legacyresample->sinkcaps, NULL);
+ gst_caps_replace (&legacyresample->srccaps, NULL);
return TRUE;
}
static gboolean
-audioresample_get_unit_size (GstBaseTransform * base, GstCaps * caps,
+legacyresample_get_unit_size (GstBaseTransform * base, GstCaps * caps,
guint * size)
{
gint width, channels;
@@ -261,7 +262,7 @@ audioresample_get_unit_size (GstBaseTransform * base, GstCaps * caps,
}
static GstCaps *
-audioresample_transform_caps (GstBaseTransform * base,
+legacyresample_transform_caps (GstBaseTransform * base,
GstPadDirection direction, GstCaps * caps)
{
GstCaps *res;
@@ -278,7 +279,7 @@ audioresample_transform_caps (GstBaseTransform * base,
/* Fixate rate to the allowed rate that has the smallest difference */
static void
-audioresample_fixate_caps (GstBaseTransform * base,
+legacyresample_fixate_caps (GstBaseTransform * base,
GstPadDirection direction, GstCaps * caps, GstCaps * othercaps)
{
GstStructure *s;
@@ -387,11 +388,11 @@ no_out_rate:
}
static gboolean
-audioresample_transform_size (GstBaseTransform * base,
+legacyresample_transform_size (GstBaseTransform * base,
GstPadDirection direction, GstCaps * caps, guint size, GstCaps * othercaps,
guint * othersize)
{
- GstAudioresample *audioresample = GST_AUDIORESAMPLE (base);
+ GstLegacyresample *legacyresample = GST_LEGACYRESAMPLE (base);
ResampleState *state;
GstCaps *srccaps, *sinkcaps;
gboolean use_internal = FALSE; /* whether we use the internal state */
@@ -409,15 +410,15 @@ audioresample_transform_size (GstBaseTransform * base,
/* 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)) {
+ if (gst_caps_is_equal (sinkcaps, legacyresample->sinkcaps) &&
+ gst_caps_is_equal (srccaps, legacyresample->srccaps)) {
use_internal = TRUE;
- state = audioresample->resample;
+ state = legacyresample->resample;
} else {
- GST_DEBUG_OBJECT (audioresample,
+ GST_DEBUG_OBJECT (legacyresample,
"caps are not the set caps, creating state");
state = resample_new ();
- resample_set_filter_length (state, audioresample->filter_length);
+ resample_set_filter_length (state, legacyresample->filter_length);
resample_set_state_from_caps (state, sinkcaps, srccaps, NULL, NULL, NULL);
}
@@ -442,64 +443,64 @@ audioresample_transform_size (GstBaseTransform * base,
}
static gboolean
-audioresample_set_caps (GstBaseTransform * base, GstCaps * incaps,
+legacyresample_set_caps (GstBaseTransform * base, GstCaps * incaps,
GstCaps * outcaps)
{
gboolean ret;
gint inrate, outrate;
int channels;
- GstAudioresample *audioresample = GST_AUDIORESAMPLE (base);
+ GstLegacyresample *legacyresample = GST_LEGACYRESAMPLE (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,
+ ret = resample_set_state_from_caps (legacyresample->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);
+ legacyresample->channels = channels;
+ GST_DEBUG_OBJECT (legacyresample, "set channels to %d", channels);
+ legacyresample->i_rate = inrate;
+ GST_DEBUG_OBJECT (legacyresample, "set i_rate to %d", inrate);
+ legacyresample->o_rate = outrate;
+ GST_DEBUG_OBJECT (legacyresample, "set o_rate to %d", outrate);
/* save caps so we can short-circuit in the size_transform if the caps
* are the same */
- gst_caps_replace (&audioresample->sinkcaps, incaps);
- gst_caps_replace (&audioresample->srccaps, outcaps);
+ gst_caps_replace (&legacyresample->sinkcaps, incaps);
+ gst_caps_replace (&legacyresample->srccaps, outcaps);
return TRUE;
}
static gboolean
-audioresample_event (GstBaseTransform * base, GstEvent * event)
+legacyresample_event (GstBaseTransform * base, GstEvent * event)
{
- GstAudioresample *audioresample;
+ GstLegacyresample *legacyresample;
- audioresample = GST_AUDIORESAMPLE (base);
+ legacyresample = GST_LEGACYRESAMPLE (base);
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_FLUSH_START:
break;
case GST_EVENT_FLUSH_STOP:
- if (audioresample->resample)
- resample_input_flush (audioresample->resample);
- audioresample->ts_offset = -1;
- audioresample->next_ts = -1;
- audioresample->offset = -1;
+ if (legacyresample->resample)
+ resample_input_flush (legacyresample->resample);
+ legacyresample->ts_offset = -1;
+ legacyresample->next_ts = -1;
+ legacyresample->offset = -1;
break;
case GST_EVENT_NEWSEGMENT:
- resample_input_pushthrough (audioresample->resample);
- audioresample_pushthrough (audioresample);
- audioresample->ts_offset = -1;
- audioresample->next_ts = -1;
- audioresample->offset = -1;
+ resample_input_pushthrough (legacyresample->resample);
+ legacyresample_pushthrough (legacyresample);
+ legacyresample->ts_offset = -1;
+ legacyresample->next_ts = -1;
+ legacyresample->offset = -1;
break;
case GST_EVENT_EOS:
- resample_input_eos (audioresample->resample);
- audioresample_pushthrough (audioresample);
+ resample_input_eos (legacyresample->resample);
+ legacyresample_pushthrough (legacyresample);
break;
default:
break;
@@ -508,57 +509,59 @@ audioresample_event (GstBaseTransform * base, GstEvent * event)
}
static GstFlowReturn
-audioresample_do_output (GstAudioresample * audioresample, GstBuffer * outbuf)
+legacyresample_do_output (GstLegacyresample * legacyresample,
+ GstBuffer * outbuf)
{
int outsize;
int outsamples;
ResampleState *r;
- r = audioresample->resample;
+ r = legacyresample->resample;
outsize = resample_get_output_size (r);
- GST_LOG_OBJECT (audioresample, "audioresample can give me %d bytes", outsize);
+ GST_LOG_OBJECT (legacyresample, "legacyresample can give me %d bytes",
+ outsize);
/* protect against mem corruption */
if (outsize > GST_BUFFER_SIZE (outbuf)) {
- GST_WARNING_OBJECT (audioresample,
- "overriding audioresample's outsize %d with outbuffer's size %d",
+ GST_WARNING_OBJECT (legacyresample,
+ "overriding legacyresample's outsize %d with outbuffer's size %d",
outsize, GST_BUFFER_SIZE (outbuf));
outsize = GST_BUFFER_SIZE (outbuf);
}
/* catch possibly wrong size differences */
if (GST_BUFFER_SIZE (outbuf) - outsize > r->sample_size) {
- GST_WARNING_OBJECT (audioresample,
- "audioresample's outsize %d too far from outbuffer's size %d",
+ GST_WARNING_OBJECT (legacyresample,
+ "legacyresample's outsize %d too far from outbuffer's size %d",
outsize, GST_BUFFER_SIZE (outbuf));
}
outsize = resample_get_output_data (r, GST_BUFFER_DATA (outbuf), outsize);
outsamples = outsize / r->sample_size;
- GST_LOG_OBJECT (audioresample, "resample gave me %d bytes or %d samples",
+ GST_LOG_OBJECT (legacyresample, "resample gave me %d bytes or %d samples",
outsize, outsamples);
- GST_BUFFER_OFFSET (outbuf) = audioresample->offset;
- GST_BUFFER_TIMESTAMP (outbuf) = audioresample->next_ts;
+ GST_BUFFER_OFFSET (outbuf) = legacyresample->offset;
+ GST_BUFFER_TIMESTAMP (outbuf) = legacyresample->next_ts;
- if (audioresample->ts_offset != -1) {
- audioresample->offset += outsamples;
- audioresample->ts_offset += outsamples;
- audioresample->next_ts =
- gst_util_uint64_scale_int (audioresample->ts_offset, GST_SECOND,
- audioresample->o_rate);
- GST_BUFFER_OFFSET_END (outbuf) = audioresample->offset;
+ if (legacyresample->ts_offset != -1) {
+ legacyresample->offset += outsamples;
+ legacyresample->ts_offset += outsamples;
+ legacyresample->next_ts =
+ gst_util_uint64_scale_int (legacyresample->ts_offset, GST_SECOND,
+ legacyresample->o_rate);
+ GST_BUFFER_OFFSET_END (outbuf) = legacyresample->offset;
/* we calculate DURATION as the difference between "next" timestamp
* and current timestamp so we ensure a contiguous stream, instead of
* having rounding errors. */
- GST_BUFFER_DURATION (outbuf) = audioresample->next_ts -
+ GST_BUFFER_DURATION (outbuf) = legacyresample->next_ts -
GST_BUFFER_TIMESTAMP (outbuf);
} else {
/* no valid offset know, we can still sortof calculate the duration though */
GST_BUFFER_DURATION (outbuf) =
gst_util_uint64_scale_int (outsamples, GST_SECOND,
- audioresample->o_rate);
+ legacyresample->o_rate);
}
/* check for possible mem corruption */
@@ -566,28 +569,28 @@ audioresample_do_output (GstAudioresample * audioresample, GstBuffer * outbuf)
/* this is an error that when it happens, would need fixing in the
* resample library; we told it we wanted only GST_BUFFER_SIZE (outbuf),
* and it gave us more ! */
- GST_WARNING_OBJECT (audioresample,
- "audioresample, you memory corrupting bastard. "
+ GST_WARNING_OBJECT (legacyresample,
+ "legacyresample, you memory corrupting bastard. "
"you gave me outsize %d while my buffer was size %d",
outsize, GST_BUFFER_SIZE (outbuf));
return GST_FLOW_ERROR;
}
/* catch possibly wrong size differences */
if (GST_BUFFER_SIZE (outbuf) - outsize > r->sample_size) {
- GST_WARNING_OBJECT (audioresample,
- "audioresample's written outsize %d too far from outbuffer's size %d",
+ GST_WARNING_OBJECT (legacyresample,
+ "legacyresample's written outsize %d too far from outbuffer's size %d",
outsize, GST_BUFFER_SIZE (outbuf));
}
GST_BUFFER_SIZE (outbuf) = outsize;
- if (G_UNLIKELY (audioresample->need_discont)) {
- GST_DEBUG_OBJECT (audioresample,
+ if (G_UNLIKELY (legacyresample->need_discont)) {
+ GST_DEBUG_OBJECT (legacyresample,
"marking this buffer with the DISCONT flag");
GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
- audioresample->need_discont = FALSE;
+ legacyresample->need_discont = FALSE;
}
- GST_LOG_OBJECT (audioresample, "transformed to buffer of %d bytes, ts %"
+ GST_LOG_OBJECT (legacyresample, "transformed to buffer of %d bytes, ts %"
GST_TIME_FORMAT ", duration %" GST_TIME_FORMAT ", offset %"
G_GINT64_FORMAT ", offset_end %" G_GINT64_FORMAT,
outsize, GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf)),
@@ -599,22 +602,22 @@ audioresample_do_output (GstAudioresample * audioresample, GstBuffer * outbuf)
}
static gboolean
-audioresample_check_discont (GstAudioresample * audioresample,
+legacyresample_check_discont (GstLegacyresample * legacyresample,
GstClockTime timestamp)
{
if (timestamp != GST_CLOCK_TIME_NONE &&
- audioresample->prev_ts != GST_CLOCK_TIME_NONE &&
- audioresample->prev_duration != GST_CLOCK_TIME_NONE &&
- timestamp != audioresample->prev_ts + audioresample->prev_duration) {
+ legacyresample->prev_ts != GST_CLOCK_TIME_NONE &&
+ legacyresample->prev_duration != GST_CLOCK_TIME_NONE &&
+ timestamp != legacyresample->prev_ts + legacyresample->prev_duration) {
/* Potentially a discontinuous buffer. However, it turns out that many
* elements generate imperfect streams due to rounding errors, so we permit
* a small error (up to one sample) without triggering a filter
* flush/restart (if triggered incorrectly, this will be audible) */
GstClockTimeDiff diff = timestamp -
- (audioresample->prev_ts + audioresample->prev_duration);
+ (legacyresample->prev_ts + legacyresample->prev_duration);
- if (ABS (diff) > GST_SECOND / audioresample->i_rate) {
- GST_WARNING_OBJECT (audioresample,
+ if (ABS (diff) > GST_SECOND / legacyresample->i_rate) {
+ GST_WARNING_OBJECT (legacyresample,
"encountered timestamp discontinuity of %" G_GINT64_FORMAT, diff);
return TRUE;
}
@@ -624,23 +627,23 @@ audioresample_check_discont (GstAudioresample * audioresample,
}
static GstFlowReturn
-audioresample_transform (GstBaseTransform * base, GstBuffer * inbuf,
+legacyresample_transform (GstBaseTransform * base, GstBuffer * inbuf,
GstBuffer * outbuf)
{
- GstAudioresample *audioresample;
+ GstLegacyresample *legacyresample;
ResampleState *r;
guchar *data, *datacopy;
gulong size;
GstClockTime timestamp;
- audioresample = GST_AUDIORESAMPLE (base);
- r = audioresample->resample;
+ legacyresample = GST_LEGACYRESAMPLE (base);
+ r = legacyresample->resample;
data = GST_BUFFER_DATA (inbuf);
size = GST_BUFFER_SIZE (inbuf);
timestamp = GST_BUFFER_TIMESTAMP (inbuf);
- GST_LOG_OBJECT (audioresample, "transforming buffer of %ld bytes, ts %"
+ GST_LOG_OBJECT (legacyresample, "transforming buffer of %ld bytes, ts %"
GST_TIME_FORMAT ", duration %" GST_TIME_FORMAT ", offset %"
G_GINT64_FORMAT ", offset_end %" G_GINT64_FORMAT,
size, GST_TIME_ARGS (timestamp),
@@ -648,16 +651,16 @@ audioresample_transform (GstBaseTransform * base, GstBuffer * inbuf,
GST_BUFFER_OFFSET (inbuf), GST_BUFFER_OFFSET_END (inbuf));
/* check for timestamp discontinuities and flush/reset if needed */
- if (G_UNLIKELY (audioresample_check_discont (audioresample, timestamp))) {
+ if (G_UNLIKELY (legacyresample_check_discont (legacyresample, timestamp))) {
/* Flush internal samples */
- audioresample_pushthrough (audioresample);
+ legacyresample_pushthrough (legacyresample);
/* Inform downstream element about discontinuity */
- audioresample->need_discont = TRUE;
+ legacyresample->need_discont = TRUE;
/* We want to recalculate the offset */
- audioresample->ts_offset = -1;
+ legacyresample->ts_offset = -1;
}
- if (audioresample->ts_offset == -1) {
+ if (legacyresample->ts_offset == -1) {
/* if we don't know the initial offset yet, calculate it based on the
* input timestamp. */
if (GST_CLOCK_TIME_IS_VALID (timestamp)) {
@@ -666,29 +669,29 @@ audioresample_transform (GstBaseTransform * base, GstBuffer * inbuf,
/* offset used to calculate the timestamps. We use the sample offset for
* this to make it more accurate. We want the first buffer to have the
* same timestamp as the incoming timestamp. */
- audioresample->next_ts = timestamp;
- audioresample->ts_offset =
+ legacyresample->next_ts = timestamp;
+ legacyresample->ts_offset =
gst_util_uint64_scale_int (timestamp, r->o_rate, GST_SECOND);
/* offset used to set as the buffer offset, this offset is always
* relative to the stream time, note that timestamp is not... */
stime = (timestamp - base->segment.start) + base->segment.time;
- audioresample->offset =
+ legacyresample->offset =
gst_util_uint64_scale_int (stime, r->o_rate, GST_SECOND);
}
}
- audioresample->prev_ts = timestamp;
- audioresample->prev_duration = GST_BUFFER_DURATION (inbuf);
+ legacyresample->prev_ts = timestamp;
+ legacyresample->prev_duration = GST_BUFFER_DURATION (inbuf);
/* need to memdup, resample takes ownership. */
datacopy = g_memdup (data, size);
resample_add_input_data (r, datacopy, size, g_free, datacopy);
- return audioresample_do_output (audioresample, outbuf);
+ return legacyresample_do_output (legacyresample, outbuf);
}
/* push remaining data in the buffers out */
static GstFlowReturn
-audioresample_pushthrough (GstAudioresample * audioresample)
+legacyresample_pushthrough (GstLegacyresample * legacyresample)
{
int outsize;
ResampleState *r;
@@ -696,25 +699,25 @@ audioresample_pushthrough (GstAudioresample * audioresample)
GstFlowReturn res = GST_FLOW_OK;
GstBaseTransform *trans;
- r = audioresample->resample;
+ r = legacyresample->resample;
outsize = resample_get_output_size (r);
if (outsize == 0) {
- GST_DEBUG_OBJECT (audioresample, "no internal buffers needing flush");
+ GST_DEBUG_OBJECT (legacyresample, "no internal buffers needing flush");
goto done;
}
- trans = GST_BASE_TRANSFORM (audioresample);
+ trans = GST_BASE_TRANSFORM (legacyresample);
res = gst_pad_alloc_buffer (trans->srcpad, GST_BUFFER_OFFSET_NONE, outsize,
GST_PAD_CAPS (trans->srcpad), &outbuf);
if (G_UNLIKELY (res != GST_FLOW_OK)) {
- GST_WARNING_OBJECT (audioresample, "failed allocating buffer of %d bytes",
+ GST_WARNING_OBJECT (legacyresample, "failed allocating buffer of %d bytes",
outsize);
goto done;
}
- res = audioresample_do_output (audioresample, outbuf);
+ res = legacyresample_do_output (legacyresample, outbuf);
if (G_UNLIKELY (res != GST_FLOW_OK))
goto done;
@@ -725,11 +728,11 @@ done:
}
static gboolean
-audioresample_query (GstPad * pad, GstQuery * query)
+legacyresample_query (GstPad * pad, GstQuery * query)
{
- GstAudioresample *audioresample =
- GST_AUDIORESAMPLE (gst_pad_get_parent (pad));
- GstBaseTransform *trans = GST_BASE_TRANSFORM (audioresample);
+ GstLegacyresample *legacyresample =
+ GST_LEGACYRESAMPLE (gst_pad_get_parent (pad));
+ GstBaseTransform *trans = GST_BASE_TRANSFORM (legacyresample);
gboolean res = TRUE;
switch (GST_QUERY_TYPE (query)) {
@@ -739,8 +742,8 @@ audioresample_query (GstPad * pad, GstQuery * query)
gboolean live;
guint64 latency;
GstPad *peer;
- gint rate = audioresample->i_rate;
- gint resampler_latency = audioresample->filter_length / 2;
+ gint rate = legacyresample->i_rate;
+ gint resampler_latency = legacyresample->filter_length / 2;
if (gst_base_transform_is_passthrough (trans))
resampler_latency = 0;
@@ -780,12 +783,12 @@ audioresample_query (GstPad * pad, GstQuery * query)
res = gst_pad_query_default (pad, query);
break;
}
- gst_object_unref (audioresample);
+ gst_object_unref (legacyresample);
return res;
}
static const GstQueryType *
-audioresample_query_type (GstPad * pad)
+legacyresample_query_type (GstPad * pad)
{
static const GstQueryType types[] = {
GST_QUERY_LATENCY,
@@ -796,23 +799,23 @@ audioresample_query_type (GstPad * pad)
}
static void
-gst_audioresample_set_property (GObject * object, guint prop_id,
+gst_legacyresample_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
- GstAudioresample *audioresample;
+ GstLegacyresample *legacyresample;
- audioresample = GST_AUDIORESAMPLE (object);
+ legacyresample = GST_LEGACYRESAMPLE (object);
switch (prop_id) {
case PROP_FILTERLEN:
- audioresample->filter_length = g_value_get_int (value);
- GST_DEBUG_OBJECT (GST_ELEMENT (audioresample), "new filter length %d",
- audioresample->filter_length);
- if (audioresample->resample) {
- resample_set_filter_length (audioresample->resample,
- audioresample->filter_length);
- gst_element_post_message (GST_ELEMENT (audioresample),
- gst_message_new_latency (GST_OBJECT (audioresample)));
+ legacyresample->filter_length = g_value_get_int (value);
+ GST_DEBUG_OBJECT (GST_ELEMENT (legacyresample), "new filter length %d",
+ legacyresample->filter_length);
+ if (legacyresample->resample) {
+ resample_set_filter_length (legacyresample->resample,
+ legacyresample->filter_length);
+ gst_element_post_message (GST_ELEMENT (legacyresample),
+ gst_message_new_latency (GST_OBJECT (legacyresample)));
}
break;
default:
@@ -822,16 +825,16 @@ gst_audioresample_set_property (GObject * object, guint prop_id,
}
static void
-gst_audioresample_get_property (GObject * object, guint prop_id,
+gst_legacyresample_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
- GstAudioresample *audioresample;
+ GstLegacyresample *legacyresample;
- audioresample = GST_AUDIORESAMPLE (object);
+ legacyresample = GST_LEGACYRESAMPLE (object);
switch (prop_id) {
case PROP_FILTERLEN:
- g_value_set_int (value, audioresample->filter_length);
+ g_value_set_int (value, legacyresample->filter_length);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
@@ -846,7 +849,7 @@ plugin_init (GstPlugin * plugin)
resample_init ();
if (!gst_element_register (plugin, "legacyresample", GST_RANK_MARGINAL,
- GST_TYPE_AUDIORESAMPLE)) {
+ GST_TYPE_LEGACYRESAMPLE)) {
return FALSE;
}
diff --git a/gst/audioresample/gstaudioresample.h b/gst/legacyresample/gstlegacyresample.h
index c969ccdb..59babc17 100644
--- a/gst/audioresample/gstaudioresample.h
+++ b/gst/legacyresample/gstlegacyresample.h
@@ -18,8 +18,8 @@
*/
-#ifndef __AUDIORESAMPLE_H__
-#define __AUDIORESAMPLE_H__
+#ifndef __LEGACYRESAMPLE_H__
+#define __LEGACYRESAMPLE_H__
#include <gst/gst.h>
#include <gst/base/gstbasetransform.h>
@@ -28,26 +28,26 @@
G_BEGIN_DECLS
-#define GST_TYPE_AUDIORESAMPLE \
- (gst_audioresample_get_type())
-#define GST_AUDIORESAMPLE(obj) \
- (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIORESAMPLE,GstAudioresample))
-#define GST_AUDIORESAMPLE_CLASS(klass) \
- (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_AUDIORESAMPLE,GstAudioresampleClass))
-#define GST_IS_AUDIORESAMPLE(obj) \
- (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIORESAMPLE))
-#define GST_IS_AUDIORESAMPLE_CLASS(klass) \
- (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_AUDIORESAMPLE))
+#define GST_TYPE_LEGACYRESAMPLE \
+ (gst_legacyresample_get_type())
+#define GST_LEGACYRESAMPLE(obj) \
+ (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_LEGACYRESAMPLE,GstLegacyresample))
+#define GST_LEGACYRESAMPLE_CLASS(klass) \
+ (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_LEGACYRESAMPLE,GstLegacyresampleClass))
+#define GST_IS_LEGACYRESAMPLE(obj) \
+ (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_LEGACYRESAMPLE))
+#define GST_IS_LEGACYRESAMPLE_CLASS(klass) \
+ (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_LEGACYRESAMPLE))
-typedef struct _GstAudioresample GstAudioresample;
-typedef struct _GstAudioresampleClass GstAudioresampleClass;
+typedef struct _GstLegacyresample GstLegacyresample;
+typedef struct _GstLegacyresampleClass GstLegacyresampleClass;
/**
- * GstAudioresample:
+ * GstLegacyresample:
*
* Opaque data structure.
*/
-struct _GstAudioresample {
+struct _GstLegacyresample {
GstBaseTransform element;
GstCaps *srccaps, *sinkcaps;
@@ -68,12 +68,12 @@ struct _GstAudioresample {
ResampleState * resample;
};
-struct _GstAudioresampleClass {
+struct _GstLegacyresampleClass {
GstBaseTransformClass parent_class;
};
-GType gst_audioresample_get_type(void);
+GType gst_legacyresample_get_type(void);
G_END_DECLS
-#endif /* __AUDIORESAMPLE_H__ */
+#endif /* __LEGACYRESAMPLE_H__ */
diff --git a/gst/audioresample/resample.c b/gst/legacyresample/resample.c
index c464adf8..c464adf8 100644
--- a/gst/audioresample/resample.c
+++ b/gst/legacyresample/resample.c
diff --git a/gst/audioresample/resample.h b/gst/legacyresample/resample.h
index 84bf8f09..84bf8f09 100644
--- a/gst/audioresample/resample.h
+++ b/gst/legacyresample/resample.h
diff --git a/gst/audioresample/resample_chunk.c b/gst/legacyresample/resample_chunk.c
index 1cf9f09f..1cf9f09f 100644
--- a/gst/audioresample/resample_chunk.c
+++ b/gst/legacyresample/resample_chunk.c
diff --git a/gst/audioresample/resample_functable.c b/gst/legacyresample/resample_functable.c
index af124276..af124276 100644
--- a/gst/audioresample/resample_functable.c
+++ b/gst/legacyresample/resample_functable.c
diff --git a/gst/audioresample/resample_ref.c b/gst/legacyresample/resample_ref.c
index bb8d2411..bb8d2411 100644
--- a/gst/audioresample/resample_ref.c
+++ b/gst/legacyresample/resample_ref.c