gst/audioconvert/bufferframesconvert.c: New element to convert buffer-frames for float streams. Not working nicely yet.

Original commit message from CVS:
2004-04-09  Andy Wingo  <wingo@pobox.com>

* gst/audioconvert/bufferframesconvert.c: New element to convert
buffer-frames for float streams. Not working nicely yet.
* gst/audioconvert/plugin.h:
* gst/audioconvert/plugin.c: New files.
* gst/audioconvert/Makefile.am: Build the new files.
* gst/audioconvert/gstaudioconvert.c: Initialize via plugin.[ch].

gst/audioconvert/Makefile.am

@@ -1,7 +1,7 @@
 
 plugin_LTLIBRARIES = libgstaudioconvert.la
 
-libgstaudioconvert_la_SOURCES = gstaudioconvert.c
+libgstaudioconvert_la_SOURCES = gstaudioconvert.c bufferframesconvert.c plugin.c plugin.h
 libgstaudioconvert_la_CFLAGS = $(GST_CFLAGS)
 libgstaudioconvert_la_LIBADD =
 libgstaudioconvert_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)

gst/audioconvert/bufferframesconvert.c

@@ -0,0 +1,349 @@
+/* GStreamer buffer-frames conversion plugin
+ * Copyright (C) 2004 Andy Wingo <wingo at pobox dot com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
+ * Boston, MA 02111-1307, USA.
+ */
+
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+
+#include <string.h>
+#include <gst/gst.h>
+#include "plugin.h"
+
+#define GSTPLUGIN_TYPE_BUFFER_FRAMES_CONVERT \
+  (gstplugin_buffer_frames_convert_get_type())
+#define BUFFER_FRAMES_CONVERT(obj) \
+  (G_TYPE_CHECK_INSTANCE_CAST ((obj), GSTPLUGIN_TYPE_BUFFER_FRAMES_CONVERT, BufferFramesConvert))
+
+typedef struct _BufferFramesConvert BufferFramesConvert;
+typedef struct _BufferFramesConvertClass BufferFramesConvertClass;
+
+struct _BufferFramesConvert
+{
+  GstElement element;
+
+  GstPad *sinkpad;
+  GstPad *srcpad;
+
+  gint in_buffer_samples;
+  gint out_buffer_samples;
+
+  gboolean passthrough;
+  GstBuffer *buf_out;
+  gint64 offset;
+  gint samples_out_remaining;
+};
+
+struct _BufferFramesConvertClass
+{
+  GstElementClass parent_class;
+};
+
+static GstElementDetails details = {
+  "buffer-frames conversion",
+  "Filter/Converter/Audio",
+  "Convert between different values of the buffer-frames stream property",
+  "Andy Wingo <wingo at pobox.com>"
+};
+
+static GstStaticPadTemplate sink_static_template =
+GST_STATIC_PAD_TEMPLATE ("sink",
+    GST_PAD_SINK,
+    GST_PAD_ALWAYS,
+    GST_STATIC_CAPS ("audio/x-raw-float,"
+        "rate = (int) [ 1, MAX ],"
+        "channels = (int) [ 1, MAX ],"
+        "endianness = (int) BYTE_ORDER,"
+        "width = (int) 32," "buffer-frames = (int) [ 0, MAX ]")
+    );
+
+static GstStaticPadTemplate src_static_template =
+GST_STATIC_PAD_TEMPLATE ("src",
+    GST_PAD_SRC,
+    GST_PAD_ALWAYS,
+    GST_STATIC_CAPS ("audio/x-raw-float,"
+        "rate = (int) [ 1, MAX ],"
+        "channels = (int) [ 1, MAX ],"
+        "endianness = (int) BYTE_ORDER,"
+        "width = (int) 32," "buffer-frames = (int) [ 0, MAX ]")
+    );
+
+static void buffer_frames_convert_class_init (BufferFramesConvertClass * klass);
+static void buffer_frames_convert_base_init (BufferFramesConvertClass * klass);
+static void buffer_frames_convert_init (BufferFramesConvert * this);
+
+static GstElementStateReturn buffer_frames_convert_change_state (GstElement *
+    element);
+
+static GstCaps *buffer_frames_convert_getcaps (GstPad * pad);
+static GstPadLinkReturn buffer_frames_convert_link (GstPad * pad,
+    const GstCaps * caps);
+
+static void buffer_frames_convert_chain (GstPad * sinkpad, GstData * _data);
+
+static GstElementClass *parent_class = NULL;
+
+GType
+gstplugin_buffer_frames_convert_get_type (void)
+{
+  static GType buffer_frames_convert_type = 0;
+
+  if (!buffer_frames_convert_type) {
+    static const GTypeInfo buffer_frames_convert_info = {
+      sizeof (BufferFramesConvertClass),
+      (GBaseInitFunc) buffer_frames_convert_base_init,
+      NULL,
+      (GClassInitFunc) buffer_frames_convert_class_init,
+      NULL,
+      NULL,
+      sizeof (BufferFramesConvert),
+      0,
+      (GInstanceInitFunc) buffer_frames_convert_init,
+    };
+
+    buffer_frames_convert_type =
+        g_type_register_static (GST_TYPE_ELEMENT, "GstBufferFramesConvert",
+        &buffer_frames_convert_info, 0);
+  }
+  return buffer_frames_convert_type;
+}
+
+static void
+buffer_frames_convert_base_init (BufferFramesConvertClass * klass)
+{
+  GstElementClass *eclass = GST_ELEMENT_CLASS (klass);
+
+  gst_element_class_set_details (eclass, &details);
+  gst_element_class_add_pad_template
+      (eclass, gst_static_pad_template_get (&src_static_template));
+  gst_element_class_add_pad_template
+      (eclass, gst_static_pad_template_get (&sink_static_template));
+}
+
+static void
+buffer_frames_convert_class_init (BufferFramesConvertClass * klass)
+{
+  GstElementClass *eclass = (GstElementClass *) klass;
+
+  parent_class = g_type_class_ref (GST_TYPE_ELEMENT);
+
+  eclass->change_state = buffer_frames_convert_change_state;
+}
+
+static void
+buffer_frames_convert_init (BufferFramesConvert * this)
+{
+  this->sinkpad = gst_pad_new_from_template
+      (gst_static_pad_template_get (&sink_static_template), "sink");
+  gst_element_add_pad (GST_ELEMENT (this), this->sinkpad);
+  gst_pad_set_link_function (this->sinkpad, buffer_frames_convert_link);
+  gst_pad_set_getcaps_function (this->sinkpad, buffer_frames_convert_getcaps);
+  gst_pad_set_chain_function (this->sinkpad, buffer_frames_convert_chain);
+
+  this->srcpad = gst_pad_new_from_template
+      (gst_static_pad_template_get (&src_static_template), "src");
+  gst_element_add_pad (GST_ELEMENT (this), this->srcpad);
+  gst_pad_set_link_function (this->srcpad, buffer_frames_convert_link);
+  gst_pad_set_getcaps_function (this->srcpad, buffer_frames_convert_getcaps);
+
+  this->in_buffer_samples = -1;
+  this->out_buffer_samples = -1;
+
+  this->buf_out = NULL;
+  this->samples_out_remaining = 0;
+}
+
+static GstElementStateReturn
+buffer_frames_convert_change_state (GstElement * element)
+{
+  BufferFramesConvert *this = (BufferFramesConvert *) element;
+
+  switch (GST_STATE_TRANSITION (element)) {
+    case GST_STATE_READY_TO_PAUSED:
+      this->offset = 0;
+      break;
+
+    case GST_STATE_PAUSED_TO_READY:
+      if (this->buf_out)
+        gst_buffer_unref (this->buf_out);
+      this->buf_out = NULL;
+      this->samples_out_remaining = 0;
+      break;
+
+    default:
+      break;
+  }
+
+  if (parent_class->change_state)
+    return parent_class->change_state (element);
+  return GST_STATE_SUCCESS;
+}
+
+static GstCaps *
+buffer_frames_convert_getcaps (GstPad * pad)
+{
+  BufferFramesConvert *this;
+  GstPad *otherpad;
+  GstCaps *ret;
+  int i;
+
+  this = BUFFER_FRAMES_CONVERT (GST_OBJECT_PARENT (pad));
+
+  otherpad = pad == this->srcpad ? this->sinkpad : this->srcpad;
+  ret = gst_pad_get_allowed_caps (otherpad);
+
+  for (i = 0; i < gst_caps_get_size (ret); i++)
+    gst_structure_set (gst_caps_get_structure (ret, i),
+        "buffer-frames", GST_TYPE_INT_RANGE, 0, G_MAXINT, NULL);
+
+  GST_DEBUG ("allowed caps %" GST_PTR_FORMAT, ret);
+
+  return ret;
+}
+
+static GstPadLinkReturn
+buffer_frames_convert_link (GstPad * pad, const GstCaps * caps)
+{
+  BufferFramesConvert *this;
+  GstCaps *othercaps;
+  GstPad *otherpad;
+  GstPadLinkReturn ret;
+  GstStructure *sinkstructure, *srcstructure;
+  gint numchannels;
+
+  this = BUFFER_FRAMES_CONVERT (GST_OBJECT_PARENT (pad));
+
+  otherpad = pad == this->srcpad ? this->sinkpad : this->srcpad;
+
+  /* first try to act as a passthrough */
+  ret = gst_pad_try_set_caps (otherpad, caps);
+  if (GST_PAD_LINK_SUCCESSFUL (ret)) {
+    this->passthrough = TRUE;
+    return ret;
+  }
+
+  /* then try to set unfixed buffer-frames */
+  othercaps = gst_caps_copy (caps);
+  gst_caps_set_simple (othercaps, "buffer-frames", GST_TYPE_INT_RANGE, 0,
+      G_MAXINT, NULL);
+  ret = gst_pad_try_set_caps_nonfixed (otherpad, othercaps);
+  if (GST_PAD_LINK_FAILED (ret))
+    return ret;
+
+  /* it's ok, let's record our data */
+  sinkstructure =
+      gst_caps_get_structure (pad == this->sinkpad ? caps : othercaps, 0);
+  srcstructure =
+      gst_caps_get_structure (pad == this->srcpad ? caps : othercaps, 0);
+  gst_structure_get_int (sinkstructure, "buffer-frames",
+      &this->in_buffer_samples);
+  gst_structure_get_int (srcstructure, "buffer-frames",
+      &this->out_buffer_samples);
+
+  gst_structure_get_int (sinkstructure, "channels", &numchannels);
+  this->in_buffer_samples *= numchannels;
+  this->out_buffer_samples *= numchannels;
+
+  if (this->out_buffer_samples == 0)
+    this->passthrough = TRUE;
+
+  return GST_PAD_LINK_OK;
+}
+
+static void
+buffer_frames_convert_chain (GstPad * pad, GstData * _data)
+{
+  BufferFramesConvert *this;
+  GstBuffer *buf_in, *buf_out;
+  gfloat *data_in;
+  gfloat *data_out;
+  gint i, samples_in, samples_in_remaining, samples_out_remaining,
+      out_buffer_samples;
+
+  this = (BufferFramesConvert *) GST_OBJECT_PARENT (pad);
+
+  if (this->passthrough) {
+    gst_pad_push (this->srcpad, _data);
+    return;
+  }
+
+  buf_in = (GstBuffer *) _data;
+  data_in = (gfloat *) GST_BUFFER_DATA (buf_in);
+  samples_in = samples_in_remaining =
+      GST_BUFFER_SIZE (buf_in) / sizeof (gfloat);
+  out_buffer_samples = this->out_buffer_samples;
+
+  /* deal with any leftover buffer */
+  if (this->buf_out) {
+    samples_out_remaining = this->samples_out_remaining;
+    buf_out = this->buf_out;
+    data_out = (gfloat *) GST_BUFFER_DATA (buf_out);
+    data_out += out_buffer_samples - samples_out_remaining;
+
+    i = MIN (samples_out_remaining, samples_in_remaining);
+    samples_in_remaining -= i;
+    samples_out_remaining -= i;
+    while (i--)
+      *(data_out++) = *(data_in++);
+
+    if (!samples_out_remaining) {
+      this->buf_out = NULL;
+      this->samples_out_remaining = 0;
+      gst_pad_push (this->srcpad, (GstData *) buf_out);
+    } else {
+      /* we used up the incoming samples, but didn't fill our buffer */
+      this->samples_out_remaining = samples_out_remaining;
+      gst_buffer_unref (buf_in);
+      return;
+    }
+  }
+
+  /* use a fast subbuffer while we can */
+  while (samples_in_remaining > out_buffer_samples) {
+    buf_out = gst_buffer_create_sub (buf_in,
+        (samples_in - samples_in_remaining) * sizeof (gfloat),
+        out_buffer_samples * sizeof (gfloat));
+    data_in += out_buffer_samples;
+    samples_in_remaining -= out_buffer_samples;
+    gst_pad_push (this->srcpad, (GstData *) buf_out);
+  }
+
+  /* if there's an event coming next, just push what we have */
+  if (this->in_buffer_samples && samples_in != this->in_buffer_samples
+      && samples_in_remaining) {
+    buf_out =
+        gst_buffer_create_sub (buf_in,
+        (samples_in - samples_in_remaining) * sizeof (gfloat),
+        samples_in_remaining * sizeof (gfloat));
+    gst_pad_push (this->srcpad, (GstData *) buf_out);
+  } else {
+    /* otherwise make a leftover buffer if it's necessary */
+    if (samples_in_remaining) {
+      buf_out =
+          gst_pad_alloc_buffer (this->srcpad, 0,
+          out_buffer_samples * sizeof (gfloat));
+      data_out = (gfloat *) GST_BUFFER_DATA (buf_out);
+      this->buf_out = buf_out;
+      this->samples_out_remaining = out_buffer_samples - samples_in_remaining;
+      while (samples_in_remaining--)
+        *(data_out++) = *(data_in++);
+    }
+  }
+
+  gst_buffer_unref (buf_in);
+}

gst/audioconvert/gstaudioconvert.c

@@ -38,6 +38,7 @@
 
 #include <gst/gst.h>
 #include <string.h>
+#include "plugin.h"
 
 GST_DEBUG_CATEGORY_STATIC (audio_convert_debug);
 #define GST_CAT_DEFAULT (audio_convert_debug)
@@ -100,7 +101,6 @@ static GstElementDetails audio_convert_details = {
 };
 
 /* type functions */
-static GType gst_audio_convert_get_type (void);
 static void gst_audio_convert_base_init (gpointer g_class);
 static void gst_audio_convert_class_init (GstAudioConvertClass * klass);
 static void gst_audio_convert_init (GstAudioConvert * audio_convert);
@@ -421,7 +421,6 @@ gst_audio_convert_link (GstPad * pad, const GstCaps * caps)
   }
 
   GST_DEBUG_OBJECT (this, "negotiated pad to %" GST_PTR_FORMAT, caps);
-  GST_DEBUG_OBJECT (this, "negotiated otherpad to %" GST_PTR_FORMAT, othercaps);
   return GST_PAD_LINK_OK;
 }
 
@@ -635,7 +634,6 @@ gst_audio_convert_buffer_to_default_format (GstAudioConvert * this,
 
     in = (gfloat *) GST_BUFFER_DATA (buf);
     out = (gint32 *) GST_BUFFER_DATA (ret);
-    /* increment `in' via the for, cause CLAMP duplicates the first arg */
     for (i = buf->size / sizeof (float); i > 0; i--) {
       *in *= 2147483647.0f + .5;
       *out = (gint32) CLAMP ((gint64) * in, -2147483648ll, 2147483647ll);
@@ -781,21 +779,3 @@ gst_audio_convert_channels (GstAudioConvert * this, GstBuffer * buf)
   gst_buffer_unref (buf);
   return ret;
 }
-
-/*** PLUGIN DETAILS ***********************************************************/
-
-static gboolean
-plugin_init (GstPlugin * plugin)
-{
-  if (!gst_element_register (plugin, "audioconvert", GST_RANK_PRIMARY,
-          GST_TYPE_AUDIO_CONVERT))
-    return FALSE;
-
-  return TRUE;
-}
-
-GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
-    GST_VERSION_MINOR,
-    "gstaudioconvert",
-    "Convert audio to different formats",
-    plugin_init, VERSION, "LGPL", GST_PACKAGE, GST_ORIGIN)

gst/audioconvert/plugin.c

@@ -0,0 +1,44 @@
+/* GStreamer audio conversion plugin
+ * Copyright (C) 2004 Andy Wingo <wingo at pobox.com>
+ *
+ * plugin.c: the stubs for the audioconvert plugin
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
+ * Boston, MA 02111-1307, USA.
+ */
+
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+
+#include "plugin.h"
+
+static gboolean
+plugin_init (GstPlugin * plugin)
+{
+  if (!gst_element_register (plugin, "audioconvert",
+          GST_RANK_PRIMARY, gst_audio_convert_get_type ()) ||
+      !gst_element_register (plugin, "buffer-frames-convert",
+          GST_RANK_NONE, gstplugin_buffer_frames_convert_get_type ()))
+    return FALSE;
+
+  return TRUE;
+}
+
+GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
+    GST_VERSION_MINOR,
+    "gstaudioconvert",
+    "Convert audio to different formats",
+    plugin_init, VERSION, "LGPL", GST_PACKAGE, GST_ORIGIN)

gst/audioconvert/plugin.h

@@ -0,0 +1,36 @@
+/* GStreamer buffer-frames conversion plugin
+ * Copyright (C) 2004 Andy Wingo <wingo at pobox.com>
+ *
+ * plugin.h: the stubs for the buffer-frames-convert plugin
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
+ * Boston, MA 02111-1307, USA.
+ */
+
+
+#ifndef __GST_PLUGIN_BUFFERFRAMESCONVERT_H__
+#define __GST_PLUGIN_BUFFERFRAMESCONVERT_H__
+
+
+#include <gst/gst.h>
+
+G_BEGIN_DECLS
+
+GType gst_audio_convert_get_type (void);
+GType gstplugin_buffer_frames_convert_get_type (void);
+
+G_END_DECLS
+
+#endif /* __GST_PLUGIN_BUFFERFRAMESCONVERT_H__ */