gstreamer/gst/audioconvert/gstaudioconvert.c

/* GStreamer
 * Copyright (C) 2003 Benjamin Otte <in7y118@public.uni-hamburg.de>
 *
 * gstaudioconvert.c: Convert audio to different audio formats automatically
 *
 * 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.
 */
/* Element-Checklist-Version: 5 */

/*
 * design decisions:
 * - audioconvert converts buffers in a set of supported caps. If it supports 
 *   a caps, it supports conversion from these caps to any other caps it 
 *   supports. (example: if it does A=>B and A=>C, it also does B=>C)
 * - audioconvert does not save state between buffers. Every incoming buffer is
 *   converted and the converted buffer is pushed out.
 * conclusion:
 * audioconvert is not supposed to be a one-element-does-anything solution for
 * audio conversions.
 */

#ifdef HAVE_CONFIG_H
#include "config.h"
#endif

#include <gst/gst.h>
#include <gst/audio/multichannel.h>
#include <string.h>
#include "gstchannelmix.h"
#include "plugin.h"

GST_DEBUG_CATEGORY (audio_convert_debug);

/*** DEFINITIONS **************************************************************/

static GstElementDetails audio_convert_details = {
  "Audio Conversion",
  "Filter/Converter/Audio",
  "Convert audio to different formats",
  "Benjamin Otte <in7y118@public.uni-hamburg.de>",
};

/* type functions */
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);
static void gst_audio_convert_dispose (GObject * obj);

/* gstreamer functions */
static GstFlowReturn gst_audio_convert_chain (GstPad * pad, GstBuffer * buffer);
static gboolean gst_audio_convert_link_src (GstAudioConvert * this,
    GstCaps * sinkcaps, GstAudioConvertCaps * sink_ac_caps);
static gboolean gst_audio_convert_setcaps (GstPad * pad, GstCaps * caps);
static void gst_audio_convert_fixate (GstPad * pad, GstCaps * caps);
static GstCaps *gst_audio_convert_getcaps (GstPad * pad);
static GstElementStateReturn gst_audio_convert_change_state (GstElement *
    element);

static GstBuffer *gst_audio_convert_buffer_to_default_format (GstAudioConvert *
    this, GstBuffer * buf);
static GstBuffer *gst_audio_convert_buffer_from_default_format (GstAudioConvert
    * this, GstBuffer * buf);

static GstBuffer *gst_audio_convert_channels (GstAudioConvert * this,
    GstBuffer * buf);

/* AudioConvert signals and args */
enum
{
  /* FILL ME */
  LAST_SIGNAL
};

enum
{
  ARG_0,
  ARG_AGGRESSIVE
};

#define DEBUG_INIT(bla) \
  GST_DEBUG_CATEGORY_INIT (audio_convert_debug, "audioconvert", 0, "audio conversion element");

GST_BOILERPLATE_FULL (GstAudioConvert, gst_audio_convert, GstElement,
    GST_TYPE_ELEMENT, DEBUG_INIT);

/*** GSTREAMER PROTOTYPES *****************************************************/

#define STATIC_CAPS \
GST_STATIC_CAPS ( \
  "audio/x-raw-float, " \
    "rate = (int) [ 1, MAX ], " \
    "channels = (int) [ 1, 8 ], " \
    "endianness = (int) BYTE_ORDER, " \
    "width = (int) 32, " \
    "buffer-frames = (int) [ 0, MAX ];" \
  "audio/x-raw-int, " \
    "rate = (int) [ 1, MAX ], " \
    "channels = (int) [ 1, 8 ], " \
    "endianness = (int) { LITTLE_ENDIAN, BIG_ENDIAN }, " \
    "width = (int) 32, " \
    "depth = (int) [ 1, 32 ], " \
    "signed = (boolean) { true, false }; " \
  "audio/x-raw-int, " \
    "rate = (int) [ 1, MAX ], " \
    "channels = (int) [ 1, 8 ], " \
    "endianness = (int) { LITTLE_ENDIAN, BIG_ENDIAN }, " \
    "width = (int) 24, " \
    "depth = (int) [ 1, 24 ], " \
    "signed = (boolean) { true, false }; " \
  "audio/x-raw-int, " \
    "rate = (int) [ 1, MAX ], " \
    "channels = (int) [ 1, 8 ], " \
    "endianness = (int) { LITTLE_ENDIAN, BIG_ENDIAN }, " \
    "width = (int) 16, " \
    "depth = (int) [ 1, 16 ], " \
    "signed = (boolean) { true, false }; " \
  "audio/x-raw-int, " \
    "rate = (int) [ 1, MAX ], " \
    "channels = (int) [ 1, 8 ], " \
    "endianness = (int) { LITTLE_ENDIAN, BIG_ENDIAN }, " \
    "width = (int) 8, " \
    "depth = (int) [ 1, 8 ], " \
    "signed = (boolean) { true, false } " \
)

static GstAudioChannelPosition *supported_positions;

static GstStaticPadTemplate gst_audio_convert_src_template =
GST_STATIC_PAD_TEMPLATE ("src",
    GST_PAD_SRC,
    GST_PAD_ALWAYS,
    STATIC_CAPS);

static GstStaticPadTemplate gst_audio_convert_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
    GST_PAD_SINK,
    GST_PAD_ALWAYS,
    STATIC_CAPS);

/*** TYPE FUNCTIONS ***********************************************************/

static void
gst_audio_convert_base_init (gpointer g_class)
{
  GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);

  gst_element_class_add_pad_template (element_class,
      gst_static_pad_template_get (&gst_audio_convert_src_template));
  gst_element_class_add_pad_template (element_class,
      gst_static_pad_template_get (&gst_audio_convert_sink_template));
  gst_element_class_set_details (element_class, &audio_convert_details);
}

static void
gst_audio_convert_class_init (GstAudioConvertClass * klass)
{
  GstElementClass *gstelement_class = GST_ELEMENT_CLASS (klass);
  GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
  gint i;

  gstelement_class->change_state = gst_audio_convert_change_state;
  gobject_class->dispose = gst_audio_convert_dispose;

  supported_positions = g_new0 (GstAudioChannelPosition,
      GST_AUDIO_CHANNEL_POSITION_NUM);
  for (i = 0; i < GST_AUDIO_CHANNEL_POSITION_NUM; i++)
    supported_positions[i] = i;
}

static void
gst_audio_convert_init (GstAudioConvert * this)
{
  /* sinkpad */
  this->sink =
      gst_pad_new_from_template (gst_static_pad_template_get
      (&gst_audio_convert_sink_template), "sink");
  gst_pad_set_getcaps_function (this->sink, gst_audio_convert_getcaps);
  gst_pad_set_setcaps_function (this->sink, gst_audio_convert_setcaps);
  gst_pad_set_fixatecaps_function (this->sink, gst_audio_convert_fixate);
  gst_element_add_pad (GST_ELEMENT (this), this->sink);

  /* srcpad */
  this->src =
      gst_pad_new_from_template (gst_static_pad_template_get
      (&gst_audio_convert_src_template), "src");
  gst_pad_set_getcaps_function (this->src, gst_audio_convert_getcaps);
  //gst_pad_set_setcaps_function (this->src, gst_audio_convert_setcaps);
  gst_pad_set_fixatecaps_function (this->src, gst_audio_convert_fixate);
  gst_element_add_pad (GST_ELEMENT (this), this->src);

  gst_pad_set_chain_function (this->sink, gst_audio_convert_chain);

  /* clear important variables */
  this->convert_internal = NULL;
  this->sinkcaps.pos = NULL;
  this->srccaps.pos = NULL;
  this->matrix = NULL;
}

static void
gst_audio_convert_dispose (GObject * obj)
{
  GstAudioConvert *this = GST_AUDIO_CONVERT (obj);

  if (this->sinkcaps.pos) {
    g_free (this->sinkcaps.pos);
    this->sinkcaps.pos = NULL;
  }

  if (this->srccaps.pos) {
    g_free (this->srccaps.pos);
    this->srccaps.pos = NULL;
  }
  G_OBJECT_CLASS (parent_class)->dispose (obj);
}

/*** GSTREAMER FUNCTIONS ******************************************************/

static GstFlowReturn
gst_audio_convert_chain (GstPad * pad, GstBuffer * buf)
{
  GstAudioConvert *this;
  GstFlowReturn ret;

  this = GST_AUDIO_CONVERT (GST_OBJECT_PARENT (pad));

  /**
   * Theory of operation:
   * - convert the format (endianness, signedness, width, depth) to
   *   (G_BYTE_ORDER, TRUE, 32, 32)
   * - convert rate and channels
   * - convert back to output format
   */
  if (!GST_PAD_CAPS (this->sink)) {
    goto not_negotiated;
  } else if (!GST_PAD_CAPS (this->src)) {
    if (!gst_audio_convert_link_src (this,
            GST_PAD_CAPS (this->sink), &this->sinkcaps))
      goto no_format;
  } else if (!this->matrix) {
    gst_audio_convert_setup_matrix (this);
  }

  buf = gst_audio_convert_buffer_to_default_format (this, buf);
  buf = gst_audio_convert_channels (this, buf);
  buf = gst_audio_convert_buffer_from_default_format (this, buf);

  ret = gst_pad_push (this->src, buf);

  return ret;

not_negotiated:
  {
    GST_ELEMENT_ERROR (this, CORE, NEGOTIATION, (NULL),
        ("Pad not negotiated before chain function was called"));
    gst_buffer_unref (buf);
    return GST_FLOW_NOT_NEGOTIATED;
  }
no_format:
  {
    GST_ELEMENT_ERROR (this, CORE, NEGOTIATION, (NULL),
        ("Could not negotiate format"));
    gst_buffer_unref (buf);
    return GST_FLOW_ERROR;
  }
}

static GstCaps *
gst_audio_convert_caps_remove_format_info (GstPad * pad, GstCaps * caps)
{
  int i, size;
  GstAudioConvert *this;

  this = GST_AUDIO_CONVERT (GST_OBJECT_PARENT (pad));

  size = gst_caps_get_size (caps);

  caps = gst_caps_make_writable (caps);

  for (i = size - 1; i >= 0; i--) {
    GstStructure *structure;

    structure = gst_caps_get_structure (caps, i);
    gst_structure_remove_field (structure, "channels");
    gst_structure_remove_field (structure, "channel-positions");
    gst_structure_remove_field (structure, "endianness");
    gst_structure_remove_field (structure, "width");
    gst_structure_remove_field (structure, "depth");
    gst_structure_remove_field (structure, "signed");
    structure = gst_structure_copy (structure);
    if (strcmp (gst_structure_get_name (structure), "audio/x-raw-int") == 0) {
      gst_structure_set_name (structure, "audio/x-raw-float");
      if (pad == this->sink) {
        gst_structure_set (structure, "buffer-frames", GST_TYPE_INT_RANGE, 0,
            G_MAXINT, NULL);
      } else {
        gst_structure_set (structure, "buffer-frames", G_TYPE_INT, 0, NULL);
      }
    } else {
      gst_structure_set_name (structure, "audio/x-raw-int");
      gst_structure_remove_field (structure, "buffer-frames");
    }
    gst_caps_append_structure (caps, structure);
  }

  return caps;
}

/* this function is complicated now, but it will be unnecessary when we convert
 * rate. */
static GstCaps *
gst_audio_convert_getcaps (GstPad * pad)
{
  GstAudioConvert *this;
  GstPad *otherpad;
  GstCaps *othercaps, *caps;
  const GstCaps *templcaps;

  this = GST_AUDIO_CONVERT (GST_OBJECT_PARENT (pad));

  otherpad = (pad == this->src) ? this->sink : this->src;

  /* we can do all our peer can */
  othercaps = gst_pad_peer_get_caps (otherpad);
  if (othercaps != NULL) {
    /* without the format info even */
    othercaps = gst_audio_convert_caps_remove_format_info (pad, othercaps);
    /* but filtered against our template */
    templcaps = gst_pad_get_pad_template_caps (pad);
    caps = gst_caps_intersect (othercaps, templcaps);
    gst_caps_unref (othercaps);
  } else {
    /* no peer, then our template is enough */
    caps = gst_caps_copy (gst_pad_get_pad_template_caps (pad));
  }

  /* Get the channel positions in as well. */
  gst_audio_set_caps_channel_positions_list (caps, supported_positions,
      GST_AUDIO_CHANNEL_POSITION_NUM);

  return caps;
}

static gboolean
gst_audio_convert_parse_caps (const GstCaps * gst_caps,
    GstAudioConvertCaps * caps)
{
  GstStructure *structure = gst_caps_get_structure (gst_caps, 0);

  GST_DEBUG ("parse caps %p and %" GST_PTR_FORMAT, gst_caps, gst_caps);

  g_return_val_if_fail (gst_caps_is_fixed (gst_caps), FALSE);
  g_return_val_if_fail (caps != NULL, FALSE);

  /* cleanup old */
  if (caps->pos) {
    g_free (caps->pos);
    caps->pos = NULL;
  }

  caps->endianness = G_BYTE_ORDER;
  caps->is_int =
      (strcmp (gst_structure_get_name (structure), "audio/x-raw-int") == 0);
  if (!gst_structure_get_int (structure, "channels", &caps->channels)
      || !(caps->pos = gst_audio_get_channel_positions (structure))
      || !gst_structure_get_int (structure, "width", &caps->width)
      || !gst_structure_get_int (structure, "rate", &caps->rate)
      || (caps->is_int
          && (!gst_structure_get_boolean (structure, "signed", &caps->sign)
              || !gst_structure_get_int (structure, "depth", &caps->depth)
              || (caps->width != 8
                  && !gst_structure_get_int (structure, "endianness",
                      &caps->endianness)))) || (!caps->is_int
          && !gst_structure_get_int (structure, "buffer-frames",
              &caps->buffer_frames))) {
    GST_DEBUG ("could not get some values from structure");
    g_free (caps->pos);
    caps->pos = NULL;
    return FALSE;
  }
  if (caps->is_int && caps->depth > caps->width) {
    GST_DEBUG ("width > depth, not allowed - make us advertise correct caps");
    g_free (caps->pos);
    caps->pos = NULL;
    return FALSE;
  }
  return TRUE;
}

static gboolean
gst_audio_convert_link_src (GstAudioConvert * this,
    GstCaps * sinkcaps, GstAudioConvertCaps * sink_ac_caps)
{
  GstAudioConvertCaps ac_caps = { 0 };

  if (gst_pad_peer_accept_caps (this->src, sinkcaps)) {
    /* great, so that will be our suggestion then */
    this->src_prefered = gst_caps_ref (sinkcaps);
    gst_caps_replace (&GST_PAD_CAPS (this->src), sinkcaps);
    ac_caps = *sink_ac_caps;
    if (ac_caps.pos) {
      ac_caps.pos = g_memdup (ac_caps.pos, sizeof (gint) * ac_caps.channels);
    }
  } else {
    /* nope, find something we can convert to and the peer can
     * accept. */
    GstCaps *othercaps = gst_pad_peer_get_caps (this->src);

    if (othercaps) {
      /* peel off first one */
      GstCaps *targetcaps = gst_caps_copy_nth (othercaps, 0);
      GstStructure *structure = gst_caps_get_structure (targetcaps, 0);

      gst_caps_unref (othercaps);

      /* set the rate on the caps, this has to work */
      gst_structure_set (structure,
          "rate", G_TYPE_INT, sink_ac_caps->rate,
          "channels", G_TYPE_INT, sink_ac_caps->channels, NULL);

      if (strcmp (gst_structure_get_name (structure), "audio/x-raw-float") == 0) {
        if (!sink_ac_caps->is_int) {
          /* copy over */
          gst_structure_set (structure, "buffer-frames", G_TYPE_INT,
              ac_caps.buffer_frames, NULL);
        } else {
          /* set to anything */
          gst_structure_set (structure, "buffer-frames", G_TYPE_INT, 0, NULL);
        }
      }

      /* this will be our suggestion */
      this->src_prefered = targetcaps;
      if (!gst_audio_convert_parse_caps (targetcaps, &ac_caps))
        return FALSE;
      gst_caps_replace (&GST_PAD_CAPS (this->src), targetcaps);
    }
  }
  this->srccaps = ac_caps;

  GST_DEBUG_OBJECT (this, "negotiated pad to %" GST_PTR_FORMAT, sinkcaps);

  return TRUE;
}

static gboolean
gst_audio_convert_setcaps (GstPad * pad, GstCaps * caps)
{
  GstAudioConvert *this;
  GstAudioConvertCaps ac_caps = { 0 };
  gboolean res;

  g_return_val_if_fail (GST_IS_PAD (pad), FALSE);
  g_return_val_if_fail (GST_IS_AUDIO_CONVERT (GST_OBJECT_PARENT (pad)), FALSE);
  g_return_val_if_fail (gst_caps_is_fixed (caps), FALSE);

  this = GST_AUDIO_CONVERT (GST_OBJECT_PARENT (pad));

  /* we'll need a new matrix after every new negotiation */
  gst_audio_convert_unset_matrix (this);

  ac_caps.pos = NULL;
  if (!gst_audio_convert_parse_caps (caps, &ac_caps))
    return FALSE;

  this->sink_prefered = caps;

  if ((res = gst_audio_convert_link_src (this, caps, &ac_caps))) {
    this->sinkcaps = ac_caps;

    GST_DEBUG_OBJECT (this, "negotiated pad to %" GST_PTR_FORMAT, caps);
  }

  return res;
}

/* tries to fixate the given field of the given caps to the given int value */
gboolean
_fixate_caps_to_int (GstCaps * caps, const gchar * field, gint value)
{
  gboolean changed = FALSE;
  guint i;

  /* FIXME: why don't we already return here when ret == TRUE ? */
  for (i = 0; i < gst_caps_get_size (caps); i++) {
    GstStructure *structure = gst_caps_get_structure (caps, i);

    if (gst_structure_has_field (structure, field))
      changed |=
          gst_caps_structure_fixate_field_nearest_int (structure, field, value);
  }
  return changed;
}

static void
gst_audio_convert_fixate (GstPad * pad, GstCaps * caps)
{
  const GValue *pos_val;
  GstPad *otherpad;
  GstAudioConvert *this;
  GstAudioConvertCaps try, ac_caps;

  this = GST_AUDIO_CONVERT (GST_PAD_PARENT (pad));
  otherpad = (pad == this->sink ? this->src : this->sink);
  ac_caps = (pad == this->sink ? this->srccaps : this->sinkcaps);

  if (!GST_PAD_IS_IN_SETCAPS (otherpad)) {
    try.channels = ac_caps.channels;
    try.width = ac_caps.is_int ? ac_caps.width : 16;
    try.depth = ac_caps.is_int ? ac_caps.depth : 16;
    try.endianness = ac_caps.is_int ? ac_caps.endianness : G_BYTE_ORDER;
  } else {
    try.channels = 2;
    try.width = 16;
    try.depth = 16;
    try.endianness = G_BYTE_ORDER;
  }

  if (_fixate_caps_to_int (caps, "channels", try.channels)) {
    int n, c;

    gst_structure_get_int (gst_caps_get_structure (caps, 0), "channels", &c);
    if (c > 2) {
      /* make sure we have a channelpositions structure or array here */
      GstStructure *str;

      for (n = 0; n < gst_caps_get_size (caps); n++) {
        str = gst_caps_get_structure (caps, n);
        if (!gst_structure_get_value (str, "channel-positions")) {
          /* first try otherpad's positions, else anything */
          if (ac_caps.pos != NULL && c == ac_caps.channels) {
            gst_audio_set_channel_positions (str, ac_caps.pos);
          } else {
            gst_audio_set_structure_channel_positions_list (str,
                supported_positions, GST_AUDIO_CHANNEL_POSITION_NUM);
            /* FIXME: fixate (else we'll be less fixed than we used to) */
          }
        }
      }
    } else {
      /* make sure we don't */
      for (n = 0; n < gst_caps_get_size (caps); n++) {
        gst_structure_remove_field (gst_caps_get_structure (caps, n),
            "channel-positions");
      }
    }
  }

  _fixate_caps_to_int (caps, "width", try.width);
  if (gst_structure_get_name (gst_caps_get_structure (caps, 0))[12] == 'i') {
    _fixate_caps_to_int (caps, "depth", try.depth);
  }
  _fixate_caps_to_int (caps, "endianness", try.endianness);
  if ((pos_val = gst_structure_get_value (gst_caps_get_structure (caps, 0),
              "channel-positions")) != NULL) {
    GstAudioChannelPosition *pos;
    const GValue *pos_val_entry;
    gint i;

    for (i = 0; i < gst_value_list_get_size (pos_val); i++) {
      pos_val_entry = gst_value_list_get_value (pos_val, i);
      if (G_VALUE_TYPE (pos_val_entry) == GST_TYPE_LIST) {
        /* unfixed */
        pos =
            gst_audio_fixate_channel_positions (gst_caps_get_structure (caps,
                0));
        if (pos) {
          gst_audio_set_channel_positions (gst_caps_get_structure (caps, 0),
              pos);
          g_free (pos);
          pos = NULL;
        }
      }
    }
  }
}

static GstElementStateReturn
gst_audio_convert_change_state (GstElement * element)
{
  GstElementStateReturn ret;
  GstAudioConvert *this = GST_AUDIO_CONVERT (element);
  gint transition;

  transition = GST_STATE_TRANSITION (element);

  switch (transition) {
    default:
      break;
  }

  ret = parent_class->change_state (element);

  switch (transition) {
    case GST_STATE_PAUSED_TO_READY:
      this->convert_internal = NULL;
      gst_audio_convert_unset_matrix (this);
      gst_caps_replace (&GST_PAD_CAPS (this->sink), NULL);
      gst_caps_replace (&GST_PAD_CAPS (this->src), NULL);
      break;
    default:
      break;
  }
  return ret;
}

/* return a writable buffer of size which ideally is the same as before
   - You must unref the new buffer
   - The size of the old buffer is undefined after this operation */
static GstBuffer *
gst_audio_convert_get_buffer (GstBuffer * buf, guint size)
{
  GstBuffer *ret;

  g_assert (GST_IS_BUFFER (buf));

  GST_LOG
      ("new buffer of size %u requested. Current is: data: %p - size: %u",
      size, buf->data, buf->size);
  if (buf->size >= size && gst_buffer_is_writable (buf)) {
    gst_buffer_ref (buf);
    buf->size = size;
    GST_LOG
        ("returning same buffer with adjusted values. data: %p - size: %u",
        buf->data, buf->size);
    return buf;
  } else {
    ret = gst_buffer_new_and_alloc (size);
    g_assert (ret);
    //gst_buffer_stamp (ret, buf);
    GST_LOG ("returning new buffer. data: %p - size: %u", ret->data, ret->size);
    return ret;
  }
}

static inline guint8
GUINT8_IDENTITY (guint8 x)
{
  return x;
}
static inline guint8
GINT8_IDENTITY (gint8 x)
{
  return x;
}

#define CONVERT_TO(to, from, type, sign, endianness, LE_FUNC, BE_FUNC)		\
G_STMT_START{									\
  type value;									\
  memcpy (&value, from, sizeof (type));						\
  from -= sizeof (type);							\
  value = (endianness == G_LITTLE_ENDIAN) ? LE_FUNC (value) : BE_FUNC (value);	\
  if (sign) {									\
    to = value;									\
  } else {									\
    to = (gint64) value - (1 << (sizeof (type) * 8 - 1));			\
  }										\
}G_STMT_END;

static GstBuffer *
gst_audio_convert_buffer_to_default_format (GstAudioConvert * this,
    GstBuffer * buf)
{
  GstBuffer *ret;
  gint i, count;
  gint64 cur = 0;
  gint32 write;
  gint32 *dest;
  guint8 *src;

  if (this->sinkcaps.is_int) {
    if (this->sinkcaps.width == 32 && this->sinkcaps.depth == 32 &&
        this->sinkcaps.endianness == G_BYTE_ORDER
        && this->sinkcaps.sign == TRUE)
      return buf;

    ret =
        gst_audio_convert_get_buffer (buf,
        buf->size * 32 / this->sinkcaps.width);
    gst_buffer_set_caps (ret, GST_PAD_CAPS (this->src));

    count = ret->size / 4;
    src = buf->data + (count - 1) * (this->sinkcaps.width / 8);
    dest = (gint32 *) ret->data;
    for (i = count - 1; i >= 0; i--) {
      switch (this->sinkcaps.width) {
        case 8:
          if (this->sinkcaps.sign) {
            CONVERT_TO (cur, src, gint8, this->sinkcaps.sign,
                this->sinkcaps.endianness, GINT8_IDENTITY, GINT8_IDENTITY);
          } else {
            CONVERT_TO (cur, src, guint8, this->sinkcaps.sign,
                this->sinkcaps.endianness, GUINT8_IDENTITY, GUINT8_IDENTITY);
          }
          break;
        case 16:
          if (this->sinkcaps.sign) {
            CONVERT_TO (cur, src, gint16, this->sinkcaps.sign,
                this->sinkcaps.endianness, GINT16_FROM_LE, GINT16_FROM_BE);
          } else {
            CONVERT_TO (cur, src, guint16, this->sinkcaps.sign,
                this->sinkcaps.endianness, GUINT16_FROM_LE, GUINT16_FROM_BE);
          }
          break;
        case 24:
        {
          /* Read 24-bits LE/BE into signed 64 host-endian */
          if (this->sinkcaps.endianness == G_LITTLE_ENDIAN) {
            cur = src[0] | (src[1] << 8) | (src[2] << 16);
          } else {
            cur = src[2] | (src[1] << 8) | (src[0] << 16);
          }

          /* Sign extend */
          if ((this->sinkcaps.sign)
              && (cur & (1 << (this->sinkcaps.depth - 1))))
            cur |= ((gint64) (-1)) ^ ((1 << this->sinkcaps.depth) - 1);

          src -= 3;
        }
          break;
        case 32:
          if (this->sinkcaps.sign) {
            CONVERT_TO (cur, src, gint32, this->sinkcaps.sign,
                this->sinkcaps.endianness, GINT32_FROM_LE, GINT32_FROM_BE);
          } else {
            CONVERT_TO (cur, src, guint32, this->sinkcaps.sign,
                this->sinkcaps.endianness, GUINT32_FROM_LE, GUINT32_FROM_BE);
          }
          break;
        default:
          g_assert_not_reached ();
      }
      cur = cur * ((gint64) 1 << (32 - this->sinkcaps.depth));
      cur = CLAMP (cur, -((gint64) 1 << 32), (gint64) 0x7FFFFFFF);
      write = cur;
      memcpy (&dest[i], &write, 4);
    }
  } else {
    /* float2int */
    gfloat *in;
    gint32 *out;
    float temp;

    /* should just give the same buffer, unless it's not writable -- float is
     * already 32 bits */
    ret = gst_audio_convert_get_buffer (buf, buf->size);
    gst_buffer_set_caps (ret, GST_PAD_CAPS (this->src));

    in = (gfloat *) GST_BUFFER_DATA (buf);
    out = (gint32 *) GST_BUFFER_DATA (ret);
    for (i = buf->size / sizeof (float); i > 0; i--) {
      temp = *in * 2147483647.0f + .5;
      *out = (gint32) CLAMP ((gint64) temp, -2147483648ll, 2147483647ll);
      out++;
      in++;
    }
  }

  gst_buffer_unref (buf);
  return ret;
}

#define POPULATE(out, format, be_func, le_func) G_STMT_START{			\
  format val;									\
  format* p = (format *) out;							\
  int_value >>= (32 - this->srccaps.depth);					\
  if (this->srccaps.sign) {							\
    val = (format) int_value;							\
  } else {									\
    val = (format) int_value + (1 << (this->srccaps.depth - 1));		\
  }										\
  switch (this->srccaps.endianness) {						\
    case G_LITTLE_ENDIAN:                                                      \
      val = le_func (val);                                                     \
      break;                                                                   \
    case G_BIG_ENDIAN:                                                         \
      val = be_func (val);                                                     \
      break;                                                                   \
    default:                                                                   \
      g_assert_not_reached ();                                                 \
  };                                                                           \
  *p = val;                                                                    \
  p ++;                                                                        \
  out = (guint8 *) p;                                                          \
}G_STMT_END

static GstBuffer *
gst_audio_convert_buffer_from_default_format (GstAudioConvert * this,
    GstBuffer * buf)
{
  GstBuffer *ret;
  guint count, i;
  gint32 *src;

  if (this->srccaps.is_int && this->srccaps.width == 32
      && this->srccaps.depth == 32 && this->srccaps.endianness == G_BYTE_ORDER
      && this->srccaps.sign == TRUE)
    return buf;

  if (this->srccaps.is_int) {
    guint8 *dest;

    count = buf->size / 4;      /* size is undefined after gst_audio_convert_get_buffer! */
    ret =
        gst_audio_convert_get_buffer (buf,
        buf->size * this->srccaps.width / 32);
    gst_buffer_set_caps (ret, GST_PAD_CAPS (this->src));

    dest = ret->data;
    src = (gint32 *) buf->data;

    for (i = 0; i < count; i++) {
      gint32 int_value = *src;

      src++;
      switch (this->srccaps.width) {
        case 8:
          if (this->srccaps.sign) {
            POPULATE (dest, gint8, GINT8_IDENTITY, GINT8_IDENTITY);
          } else {
            POPULATE (dest, guint8, GUINT8_IDENTITY, GUINT8_IDENTITY);
          }
          break;
        case 16:
          if (this->srccaps.sign) {
            POPULATE (dest, gint16, GINT16_TO_BE, GINT16_TO_LE);
          } else {
            POPULATE (dest, guint16, GUINT16_TO_BE, GUINT16_TO_LE);
          }
          break;
        case 24:
        {
          guint8 tmp[4];
          guint8 *tmpp = tmp;

          /* Write out big endian array */
          if (this->srccaps.sign) {
            POPULATE (tmpp, gint32, GINT32_TO_BE, GINT32_TO_BE);
          } else {
            POPULATE (tmpp, guint32, GUINT32_TO_BE, GUINT32_TO_BE);
          }

          if (this->srccaps.endianness == G_LITTLE_ENDIAN) {
            dest[2] = tmp[1];
            dest[1] = tmp[2];
            dest[0] = tmp[3];
          } else {
            memcpy (dest, tmp + 1, 3);
          }
          dest += 3;
        }
          break;
        case 32:
          if (this->srccaps.sign) {
            POPULATE (dest, gint32, GINT32_TO_BE, GINT32_TO_LE);
          } else {
            POPULATE (dest, guint32, GUINT32_TO_BE, GUINT32_TO_LE);
          }
          break;
        default:
          g_assert_not_reached ();
      }
    }
  } else {
    gfloat *dest;

    /* 1 / (2^31-1) * i */
#define INT2FLOAT(i) (4.6566128752457969e-10 * ((gfloat)i))
    count = buf->size / 4;      /* size is undefined after gst_audio_convert_get_buffer! */
    ret =
        gst_audio_convert_get_buffer (buf,
        buf->size * this->srccaps.width / 32);
    gst_buffer_set_caps (ret, GST_PAD_CAPS (this->src));

    dest = (gfloat *) ret->data;
    src = (gint32 *) buf->data;
    for (i = 0; i < count; i++) {
      *dest = (4.6566128752457969e-10 * ((gfloat) * src));
      dest++;
      src++;
    }
  }

  gst_buffer_unref (buf);
  return ret;
}

static GstBuffer *
gst_audio_convert_channels (GstAudioConvert * this, GstBuffer * buf)
{
  GstBuffer *ret;
  gint count;

  g_assert (this->matrix != NULL);

  /* check for passthrough */
  if (gst_audio_convert_passthrough (this))
    return buf;

  /* convert */
  count = GST_BUFFER_SIZE (buf) / 4 / this->sinkcaps.channels;
  ret = gst_audio_convert_get_buffer (buf, count * 4 * this->srccaps.channels);
  gst_buffer_set_caps (ret, GST_PAD_CAPS (this->src));
  gst_audio_convert_mix (this, (gint32 *) GST_BUFFER_DATA (buf),
      (gint32 *) GST_BUFFER_DATA (ret), count);
  gst_buffer_unref (buf);

  return ret;
}