/* GStreamer

 * Copyright (C) 2003 Benjamin Otte <in7y118@public.uni-hamburg.de>

 * Copyright (C) 2005 Thomas Vander Stichele <thomas at apestaart dot org>

 * Copyright (C) 2011 Wim Taymans <wim.taymans at gmail dot com>

 *

 * 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., 51 Franklin St, Fifth Floor,

 * Boston, MA 02110-1301, USA.

 */

/**

 * SECTION:element-audioconvert

 * @title: audioconvert

 *

 * Audioconvert converts raw audio buffers between various possible formats.

 * It supports integer to float conversion, width/depth conversion,

 * signedness and endianness conversion and channel transformations

 * (ie. upmixing and downmixing), as well as dithering and noise-shaping.

 *

 * ## Example launch line

 * |[

 * gst-launch-1.0 -v -m audiotestsrc ! audioconvert ! audio/x-raw,format=S8,channels=2 ! level ! fakesink silent=TRUE

 * ]|

 *  This pipeline converts audio to 8-bit.  The level element shows that

 * the output levels still match the one for a sine wave.

 * |[

 * gst-launch-1.0 -v -m uridecodebin uri=file:///path/to/audio.flac ! audioconvert ! vorbisenc ! oggmux ! filesink location=audio.ogg

 * ]|

 *  The vorbis encoder takes float audio data instead of the integer data

 * output by most other audio elements. This pipeline decodes a FLAC audio file

 * (or any other audio file for which decoders are installed) and re-encodes

 * it into an Ogg/Vorbis audio file.

 *

 * A mix matrix can be passed to audioconvert, that will govern the

 * remapping of input to output channels.

 * ## Example matrix generation code

 * To generate the matrix using code:

 *

 * |[

 * GValue v = G_VALUE_INIT;

 * GValue v2 = G_VALUE_INIT;

 * GValue v3 = G_VALUE_INIT;

 *

 * g_value_init (&v2, GST_TYPE_ARRAY);

 * g_value_init (&v3, G_TYPE_FLOAT);

 * g_value_set_float (&v3, 1);

 * gst_value_array_append_value (&v2, &v3;;

 * g_value_unset (&v3;;

 * [ Repeat for as many float as your input channels - unset and reinit v3 ]

 * g_value_init (&v, GST_TYPE_ARRAY);

 * gst_value_array_append_value (&v, &v2;;

 * g_value_unset (&v2;;

 * [ Repeat for as many v2's as your output channels - unset and reinit v2]

 * g_object_set_property (G_OBJECT (audioconvert), "mix-matrix", &v);

 * g_value_unset (&v);

 * ]|

 *

 * ## Example launch line

 * |[

 * gst-launch-1.0 audiotestsrc ! audio/x-raw, channels=4 ! audioconvert mix-matrix="<<(float)1.0, (float)0.0, (float)0.0, (float)0.0>, <(float)0.0, (float)1.0, (float)0.0, (float)0.0>>" ! audio/x-raw,channels=2 ! autoaudiosink

 * ]|

 *

 * > If an empty mix matrix is specified, a (potentially truncated)

 * > identity matrix will be generated.

 *

 * ## Example empty matrix generation code

 * |[

 * GValue v = G_VALUE_INIT;

 *

 * g_value_init (&v, GST_TYPE_ARRAY);

 * g_object_set_property (G_OBJECT (audioconvert), "mix-matrix", &v);

 * g_value_unset (&v);

 * ]|

 *

 * ## Example empty matrix launch line

 * |[

 * gst-launch-1.0 -v audiotestsrc ! audio/x-raw,channels=8 ! audioconvert mix-matrix="<>" ! audio/x-raw,channels=16,channel-mask=\(bitmask\)0x0000000000000000 ! fakesink

 * ]|

 */

/*

 * 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 <string.h>

#include "gstaudioconvert.h"

#include "plugin.h"

GST_DEBUG_CATEGORY (audio_convert_debug);

GST_DEBUG_CATEGORY_STATIC (GST_CAT_PERFORMANCE);

#define GST_CAT_DEFAULT (audio_convert_debug)

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

/* type functions */

static void gst_audio_convert_dispose (GObject * obj);

/* gstreamer functions */

static gboolean gst_audio_convert_get_unit_size (GstBaseTransform * base,

    GstCaps * caps, gsize * size);

static GstCaps *gst_audio_convert_transform_caps (GstBaseTransform * base,

    GstPadDirection direction, GstCaps * caps, GstCaps * filter);

static GstCaps *gst_audio_convert_fixate_caps (GstBaseTransform * base,

    GstPadDirection direction, GstCaps * caps, GstCaps * othercaps);

static gboolean gst_audio_convert_set_caps (GstBaseTransform * base,

    GstCaps * incaps, GstCaps * outcaps);

static GstFlowReturn gst_audio_convert_transform (GstBaseTransform * base,

    GstBuffer * inbuf, GstBuffer * outbuf);

static GstFlowReturn gst_audio_convert_transform_ip (GstBaseTransform * base,

    GstBuffer * buf);

static gboolean gst_audio_convert_transform_meta (GstBaseTransform * trans,

    GstBuffer * outbuf, GstMeta * meta, GstBuffer * inbuf);

static GstFlowReturn gst_audio_convert_submit_input_buffer (GstBaseTransform *

    base, gboolean is_discont, GstBuffer * input);

static void gst_audio_convert_set_property (GObject * object, guint prop_id,

    const GValue * value, GParamSpec * pspec);

static void gst_audio_convert_get_property (GObject * object, guint prop_id,

    GValue * value, GParamSpec * pspec);

/* AudioConvert signals and args */

enum

{

  /* FILL ME */

  LAST_SIGNAL

};

enum

{

  PROP_0,

  PROP_DITHERING,

  PROP_NOISE_SHAPING,

  PROP_MIX_MATRIX,

};

#define DEBUG_INIT \

  GST_DEBUG_CATEGORY_INIT (audio_convert_debug, "audioconvert", 0, "audio conversion element"); \

  GST_DEBUG_CATEGORY_GET (GST_CAT_PERFORMANCE, "GST_PERFORMANCE");

#define gst_audio_convert_parent_class parent_class

G_DEFINE_TYPE_WITH_CODE (GstAudioConvert, gst_audio_convert,

    GST_TYPE_BASE_TRANSFORM, DEBUG_INIT);

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

#define STATIC_CAPS \

GST_STATIC_CAPS (GST_AUDIO_CAPS_MAKE (GST_AUDIO_FORMATS_ALL) \

    ", layout = (string) interleaved")

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_class_init (GstAudioConvertClass * klass)

{

  GObjectClass *gobject_class = G_OBJECT_CLASS (klass);

  GstElementClass *element_class = GST_ELEMENT_CLASS (klass);

  GstBaseTransformClass *basetransform_class = GST_BASE_TRANSFORM_CLASS (klass);

  gobject_class->dispose = gst_audio_convert_dispose;

  gobject_class->set_property = gst_audio_convert_set_property;

  gobject_class->get_property = gst_audio_convert_get_property;

  g_object_class_install_property (gobject_class, PROP_DITHERING,

      g_param_spec_enum ("dithering", "Dithering",

          "Selects between different dithering methods.",

          GST_TYPE_AUDIO_DITHER_METHOD, GST_AUDIO_DITHER_TPDF,

          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));

  g_object_class_install_property (gobject_class, PROP_NOISE_SHAPING,

      g_param_spec_enum ("noise-shaping", "Noise shaping",

          "Selects between different noise shaping methods.",

          GST_TYPE_AUDIO_NOISE_SHAPING_METHOD, GST_AUDIO_NOISE_SHAPING_NONE,

          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));

  g_object_class_install_property (gobject_class, PROP_MIX_MATRIX,

      gst_param_spec_array ("mix-matrix",

          "Input/output channel matrix",

          "Transformation matrix for input/output channels",

          gst_param_spec_array ("matrix-rows", "rows", "rows",

              g_param_spec_float ("matrix-cols", "cols", "cols",

                  -1, 1, 0,

                  G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS),

              G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS),

          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));

  gst_element_class_add_static_pad_template (element_class,

      &gst_audio_convert_src_template);

  gst_element_class_add_static_pad_template (element_class,

      &gst_audio_convert_sink_template);

  gst_element_class_set_static_metadata (element_class, "Audio converter",

      "Filter/Converter/Audio", "Convert audio to different formats",

      "Benjamin Otte <otte@gnome.org>");

  basetransform_class->get_unit_size =

      GST_DEBUG_FUNCPTR (gst_audio_convert_get_unit_size);

  basetransform_class->transform_caps =

      GST_DEBUG_FUNCPTR (gst_audio_convert_transform_caps);

  basetransform_class->fixate_caps =

      GST_DEBUG_FUNCPTR (gst_audio_convert_fixate_caps);

  basetransform_class->set_caps =

      GST_DEBUG_FUNCPTR (gst_audio_convert_set_caps);

  basetransform_class->transform =

      GST_DEBUG_FUNCPTR (gst_audio_convert_transform);

  basetransform_class->transform_ip =

      GST_DEBUG_FUNCPTR (gst_audio_convert_transform_ip);

  basetransform_class->transform_meta =

      GST_DEBUG_FUNCPTR (gst_audio_convert_transform_meta);

  basetransform_class->submit_input_buffer =

      GST_DEBUG_FUNCPTR (gst_audio_convert_submit_input_buffer);

  basetransform_class->passthrough_on_same_caps = TRUE;

  basetransform_class->transform_ip_on_passthrough = FALSE;

}

static void

gst_audio_convert_init (GstAudioConvert * this)

{

  this->dither = GST_AUDIO_DITHER_TPDF;

  this->ns = GST_AUDIO_NOISE_SHAPING_NONE;

  g_value_init (&this->mix_matrix, GST_TYPE_ARRAY);

  gst_base_transform_set_gap_aware (GST_BASE_TRANSFORM (this), TRUE);

}

static void

gst_audio_convert_dispose (GObject * obj)

{

  GstAudioConvert *this = GST_AUDIO_CONVERT (obj);

  if (this->convert) {

    gst_audio_converter_free (this->convert);

    this->convert = NULL;

  }

  g_value_unset (&this->mix_matrix);

  G_OBJECT_CLASS (parent_class)->dispose (obj);

}

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

/* BaseTransform vmethods */

static gboolean

gst_audio_convert_get_unit_size (GstBaseTransform * base, GstCaps * caps,

    gsize * size)

{

  GstAudioInfo info;

  g_assert (size);

  if (!gst_audio_info_from_caps (&info, caps))

    goto parse_error;

  *size = info.bpf;

  GST_INFO_OBJECT (base, "unit_size = %" G_GSIZE_FORMAT, *size);

  return TRUE;

parse_error:

  {

    GST_INFO_OBJECT (base, "failed to parse caps to get unit_size");

    return FALSE;

  }

}

static gboolean

remove_format_from_structure (GstCapsFeatures * features,

    GstStructure * structure, gpointer user_data G_GNUC_UNUSED)

{

  gst_structure_remove_field (structure, "format");

  return TRUE;

}

static gboolean

remove_channels_from_structure (GstCapsFeatures * features, GstStructure * s,

    gpointer user_data)

{

  guint64 mask;

  gint channels;

  GstAudioConvert *this = GST_AUDIO_CONVERT (user_data);

  /* Only remove the channels and channel-mask for non-NONE layouts,

   * or if a mix matrix was manually specified */

  if (this->mix_matrix_was_set ||

      !gst_structure_get (s, "channel-mask", GST_TYPE_BITMASK, &mask, NULL) ||

      (mask != 0 || (gst_structure_get_int (s, "channels", &channels)

              && channels == 1))) {

    gst_structure_remove_fields (s, "channel-mask", "channels", NULL);

  }

  return TRUE;

}

static gboolean

add_other_channels_to_structure (GstCapsFeatures * features, GstStructure * s,

    gpointer user_data)

{

  gint other_channels = GPOINTER_TO_INT (user_data);

  gst_structure_set (s, "channels", G_TYPE_INT, other_channels, NULL);

  return TRUE;

}

/* The caps can be transformed into any other caps with format info removed.

 * However, we should prefer passthrough, so if passthrough is possible,

 * put it first in the list. */

static GstCaps *

gst_audio_convert_transform_caps (GstBaseTransform * btrans,

    GstPadDirection direction, GstCaps * caps, GstCaps * filter)

{

  GstCaps *tmp, *tmp2;

  GstCaps *result;

  GstAudioConvert *this = GST_AUDIO_CONVERT (btrans);

  tmp = gst_caps_copy (caps);

  gst_caps_map_in_place (tmp, remove_format_from_structure, NULL);

  gst_caps_map_in_place (tmp, remove_channels_from_structure, btrans);

  /* We can infer the required input / output channels based on the

   * matrix dimensions */

  if (gst_value_array_get_size (&this->mix_matrix)) {

    gint other_channels;

    if (direction == GST_PAD_SRC) {

      const GValue *first_row =

          gst_value_array_get_value (&this->mix_matrix, 0);

      other_channels = gst_value_array_get_size (first_row);

    } else {

      other_channels = gst_value_array_get_size (&this->mix_matrix);

    }

    gst_caps_map_in_place (tmp, add_other_channels_to_structure,

        GINT_TO_POINTER (other_channels));

  }

  if (filter) {

    tmp2 = gst_caps_intersect_full (filter, tmp, GST_CAPS_INTERSECT_FIRST);

    gst_caps_unref (tmp);

    tmp = tmp2;

  }

  result = tmp;

  GST_DEBUG_OBJECT (btrans, "transformed %" GST_PTR_FORMAT " into %"

      GST_PTR_FORMAT, caps, result);

  return result;

}

/* Count the number of bits set

 * Optimized for the common case, assuming that the number of channels

 * (i.e. bits set) is small

 */

static gint

n_bits_set (guint64 x)

{

  gint c;

  for (c = 0; x; c++)

    x &= x - 1;

  return c;

}

/* Reduce the mask to the n_chans lowest set bits

 *

 * The algorithm clears the n_chans lowest set bits and subtracts the

 * result from the original mask to get the desired mask.

 * It is optimized for the common case where n_chans is a small

 * number. In the worst case, however, it stops after 64 iterations.

 */

static guint64

find_suitable_mask (guint64 mask, gint n_chans)

{

  guint64 x = mask;

  for (; x && n_chans; n_chans--)

    x &= x - 1;

  g_assert (x || n_chans == 0);

  /* assertion fails if mask contained less bits than n_chans

   * or n_chans was < 0 */

  return mask - x;

}

static void

gst_audio_convert_fixate_format (GstBaseTransform * base, GstStructure * ins,

    GstStructure * outs)

{

  const gchar *in_format;

  const GValue *format;

  const GstAudioFormatInfo *in_info, *out_info = NULL;

  GstAudioFormatFlags in_flags, out_flags = 0;

  gint in_depth, out_depth = -1;

  gint i, len;

  in_format = gst_structure_get_string (ins, "format");

  if (!in_format)

    return;

  format = gst_structure_get_value (outs, "format");

  /* should not happen */

  if (format == NULL)

    return;

  /* nothing to fixate? */

  if (!GST_VALUE_HOLDS_LIST (format))

    return;

  in_info =

      gst_audio_format_get_info (gst_audio_format_from_string (in_format));

  if (!in_info)

    return;

  in_flags = GST_AUDIO_FORMAT_INFO_FLAGS (in_info);

  in_flags &= ~(GST_AUDIO_FORMAT_FLAG_UNPACK);

  in_flags &= ~(GST_AUDIO_FORMAT_FLAG_SIGNED);

  in_depth = GST_AUDIO_FORMAT_INFO_DEPTH (in_info);

  len = gst_value_list_get_size (format);

  for (i = 0; i < len; i++) {

    const GstAudioFormatInfo *t_info;

    GstAudioFormatFlags t_flags;

    gboolean t_flags_better;

    const GValue *val;

    const gchar *fname;

    gint t_depth;

    val = gst_value_list_get_value (format, i);

    if (!G_VALUE_HOLDS_STRING (val))

      continue;

    fname = g_value_get_string (val);

    t_info = gst_audio_format_get_info (gst_audio_format_from_string (fname));

    if (!t_info)

      continue;

    /* accept input format immediately */

    if (strcmp (fname, in_format) == 0) {

      out_info = t_info;

      break;

    }

    t_flags = GST_AUDIO_FORMAT_INFO_FLAGS (t_info);

    t_flags &= ~(GST_AUDIO_FORMAT_FLAG_UNPACK);

    t_flags &= ~(GST_AUDIO_FORMAT_FLAG_SIGNED);

    t_depth = GST_AUDIO_FORMAT_INFO_DEPTH (t_info);

    /* Any output format is better than no output format at all */

    if (!out_info) {

      out_info = t_info;

      out_depth = t_depth;

      out_flags = t_flags;

      continue;

    }

    t_flags_better = (t_flags == in_flags && out_flags != in_flags);

    if (t_depth == in_depth && (out_depth != in_depth || t_flags_better)) {

      /* Prefer to use the first format that has the same depth with the same

       * flags, and if none with the same flags exist use the first other one

       * that has the same depth */

      out_info = t_info;

      out_depth = t_depth;

      out_flags = t_flags;

    } else if (t_depth >= in_depth && (in_depth > out_depth

            || (out_depth >= in_depth && t_flags_better))) {

      /* Otherwise use the first format that has a higher depth with the same flags,

       * if none with the same flags exist use the first other one that has a higher

       * depth */

      out_info = t_info;

      out_depth = t_depth;

      out_flags = t_flags;

    } else if ((t_depth > out_depth && out_depth < in_depth)

        || (t_flags_better && out_depth == t_depth)) {

      /* Else get at least the one with the highest depth, ideally with the same flags */

      out_info = t_info;

      out_depth = t_depth;

      out_flags = t_flags;

    }

  }

  if (out_info)

    gst_structure_set (outs, "format", G_TYPE_STRING,

        GST_AUDIO_FORMAT_INFO_NAME (out_info), NULL);

}

static void

gst_audio_convert_fixate_channels (GstBaseTransform * base, GstStructure * ins,

    GstStructure * outs)

{

  gint in_chans, out_chans;

  guint64 in_mask = 0, out_mask = 0;

  gboolean has_in_mask = FALSE, has_out_mask = FALSE;

  if (!gst_structure_get_int (ins, "channels", &in_chans))

    return;                     /* this shouldn't really happen, should it? */

  if (!gst_structure_has_field (outs, "channels")) {

    /* we could try to get the implied number of channels from the layout,

     * but that seems overdoing it for a somewhat exotic corner case */

    gst_structure_remove_field (outs, "channel-mask");

    return;

  }

  /* ok, let's fixate the channels if they are not fixated yet */

  gst_structure_fixate_field_nearest_int (outs, "channels", in_chans);

  if (!gst_structure_get_int (outs, "channels", &out_chans)) {

    /* shouldn't really happen ... */

    gst_structure_remove_field (outs, "channel-mask");

    return;

  }

  /* get the channel layout of the output if any */

  has_out_mask = gst_structure_has_field (outs, "channel-mask");

  if (has_out_mask) {

    gst_structure_get (outs, "channel-mask", GST_TYPE_BITMASK, &out_mask, NULL);

  } else {

    /* channels == 1 => MONO */

    if (out_chans == 2) {

      out_mask =

          GST_AUDIO_CHANNEL_POSITION_MASK (FRONT_LEFT) |

          GST_AUDIO_CHANNEL_POSITION_MASK (FRONT_RIGHT);

      has_out_mask = TRUE;

      gst_structure_set (outs, "channel-mask", GST_TYPE_BITMASK, out_mask,

          NULL);

    }

  }

  /* get the channel layout of the input if any */

  has_in_mask = gst_structure_has_field (ins, "channel-mask");

  if (has_in_mask) {

    gst_structure_get (ins, "channel-mask", GST_TYPE_BITMASK, &in_mask, NULL);

  } else {

    /* channels == 1 => MONO */

    if (in_chans == 2) {

      in_mask =

          GST_AUDIO_CHANNEL_POSITION_MASK (FRONT_LEFT) |

          GST_AUDIO_CHANNEL_POSITION_MASK (FRONT_RIGHT);

      has_in_mask = TRUE;

    } else if (in_chans > 2)

      g_warning ("%s: Upstream caps contain no channel mask",

          GST_ELEMENT_NAME (base));

  }

  if (!has_out_mask && out_chans == 1 && (in_chans != out_chans

          || !has_in_mask))

    return;                     /* nothing to do, default layout will be assumed */

  if (in_chans == out_chans && (has_in_mask || in_chans == 1)) {

    /* same number of channels and no output layout: just use input layout */

    if (!has_out_mask) {

      /* in_chans == 1 handled above already */

      gst_structure_set (outs, "channel-mask", GST_TYPE_BITMASK, in_mask, NULL);

      return;

    }

    /* If both masks are the same we're done, this includes the NONE layout case */

    if (in_mask == out_mask)

      return;

    /* if output layout is fixed already and looks sane, we're done */

    if (n_bits_set (out_mask) == out_chans)

      return;

    if (n_bits_set (out_mask) < in_chans) {

      /* Not much we can do here, this shouldn't just happen */

      g_warning ("%s: Invalid downstream channel-mask with too few bits set",

          GST_ELEMENT_NAME (base));

    } else {

      guint64 intersection;

      /* if the output layout is not fixed, check if the output layout contains

       * the input layout */

      intersection = in_mask & out_mask;

      if (n_bits_set (intersection) >= in_chans) {

        gst_structure_set (outs, "channel-mask", GST_TYPE_BITMASK, in_mask,

            NULL);

        return;

      }

      /* output layout is not fixed and does not contain the input layout, so

       * just pick the first possibility */

      intersection = find_suitable_mask (out_mask, out_chans);

      if (intersection) {

        gst_structure_set (outs, "channel-mask", GST_TYPE_BITMASK, intersection,

            NULL);

        return;

      }

    }

    /* ... else fall back to default layout (NB: out_layout is NULL here) */

    GST_WARNING_OBJECT (base, "unexpected output channel layout");

  } else {

    guint64 intersection;

    /* number of input channels != number of output channels:

     * if this value contains a list of channel layouts (or even worse: a list

     * with another list), just pick the first value and repeat until we find a

     * channel position array or something else that's not a list; we assume

     * the input if half-way sane and don't try to fall back on other list items

     * if the first one is something unexpected or non-channel-pos-array-y */

    if (n_bits_set (out_mask) >= out_chans) {

      intersection = find_suitable_mask (out_mask, out_chans);

      gst_structure_set (outs, "channel-mask", GST_TYPE_BITMASK, intersection,

          NULL);

      return;

    }

    /* what now?! Just ignore what we're given and use default positions */

    GST_WARNING_OBJECT (base, "invalid or unexpected channel-positions");

  }

  /* missing or invalid output layout and we can't use the input layout for

   * one reason or another, so just pick a default layout (we could be smarter

   * and try to add/remove channels from the input layout, or pick a default

   * layout based on LFE-presence in input layout, but let's save that for

   * another day). For mono, no mask is required and the fallback mask is 0 */

  if (out_chans > 1

      && (out_mask = gst_audio_channel_get_fallback_mask (out_chans))) {

    GST_DEBUG_OBJECT (base, "using default channel layout as fallback");

    gst_structure_set (outs, "channel-mask", GST_TYPE_BITMASK, out_mask, NULL);

  } else if (out_chans > 1) {

    GST_ERROR_OBJECT (base, "Have no default layout for %d channels",

        out_chans);

  }

}

/* try to keep as many of the structure members the same by fixating the

 * possible ranges; this way we convert the least amount of things as possible

 */

static GstCaps *

gst_audio_convert_fixate_caps (GstBaseTransform * base,

    GstPadDirection direction, GstCaps * caps, GstCaps * othercaps)

{

  GstStructure *ins, *outs;

  GstCaps *result;

  GST_DEBUG_OBJECT (base, "trying to fixate othercaps %" GST_PTR_FORMAT

      " based on caps %" GST_PTR_FORMAT, othercaps, caps);

  result = gst_caps_intersect (othercaps, caps);

  if (gst_caps_is_empty (result)) {

    GstCaps *removed = gst_caps_copy (caps);

    if (result)

      gst_caps_unref (result);

    gst_caps_map_in_place (removed, remove_format_from_structure, NULL);

    result = gst_caps_intersect (othercaps, removed);

    gst_caps_unref (removed);

    if (gst_caps_is_empty (result)) {

      if (result)

        gst_caps_unref (result);

      result = othercaps;

    } else {

      gst_caps_unref (othercaps);

    }

  } else {

    gst_caps_unref (othercaps);

  }

  GST_DEBUG_OBJECT (base, "now fixating %" GST_PTR_FORMAT, result);

  /* fixate remaining fields */

  result = gst_caps_make_writable (result);

  ins = gst_caps_get_structure (caps, 0);

  outs = gst_caps_get_structure (result, 0);

  gst_audio_convert_fixate_channels (base, ins, outs);

  gst_audio_convert_fixate_format (base, ins, outs);

  /* fixate remaining */

  result = gst_caps_fixate (result);

  GST_DEBUG_OBJECT (base, "fixated othercaps to %" GST_PTR_FORMAT, result);

  return result;

}

static gboolean

gst_audio_convert_set_caps (GstBaseTransform * base, GstCaps * incaps,

    GstCaps * outcaps)

{

  GstAudioConvert *this = GST_AUDIO_CONVERT (base);

  GstAudioInfo in_info;

  GstAudioInfo out_info;

  gboolean in_place;

  GstStructure *config;

  GST_DEBUG_OBJECT (base, "incaps %" GST_PTR_FORMAT ", outcaps %"

      GST_PTR_FORMAT, incaps, outcaps);

  if (this->convert) {

    gst_audio_converter_free (this->convert);

    this->convert = NULL;

  }

  if (!gst_audio_info_from_caps (&in_info, incaps))

    goto invalid_in;

  if (!gst_audio_info_from_caps (&out_info, outcaps))

    goto invalid_out;

  config = gst_structure_new ("GstAudioConverterConfig",

      GST_AUDIO_CONVERTER_OPT_DITHER_METHOD, GST_TYPE_AUDIO_DITHER_METHOD,

      this->dither,

      GST_AUDIO_CONVERTER_OPT_NOISE_SHAPING_METHOD,

      GST_TYPE_AUDIO_NOISE_SHAPING_METHOD, this->ns, NULL);

  if (this->mix_matrix_was_set)

    gst_structure_set_value (config, GST_AUDIO_CONVERTER_OPT_MIX_MATRIX,

        &this->mix_matrix);

  this->convert = gst_audio_converter_new (0, &in_info, &out_info, config);

  if (this->convert == NULL)

    goto no_converter;

  in_place = gst_audio_converter_supports_inplace (this->convert);

  gst_base_transform_set_in_place (base, in_place);

  this->in_info = in_info;

  this->out_info = out_info;

  return TRUE;

  /* ERRORS */

invalid_in:

  {

    GST_ERROR_OBJECT (base, "invalid input caps");

    return FALSE;

  }

invalid_out:

  {

    GST_ERROR_OBJECT (base, "invalid output caps");

    return FALSE;

  }

no_converter:

  {

    GST_ERROR_OBJECT (base, "could not make converter");

    return FALSE;

  }

}

/* if called through gst_audio_convert_transform_ip() inbuf == outbuf */

static GstFlowReturn

gst_audio_convert_transform (GstBaseTransform * base, GstBuffer * inbuf,

    GstBuffer * outbuf)

{

  GstFlowReturn ret;

  GstAudioConvert *this = GST_AUDIO_CONVERT (base);

  GstMapInfo srcmap = { NULL, }, dstmap;

  gint insize, outsize;

  gboolean inbuf_writable;

  GstAudioConverterFlags flags;

  gsize samples;

  /* get amount of samples to convert. */

  samples = gst_buffer_get_size (inbuf) / this->in_info.bpf;

  /* get in/output sizes, to see if the buffers we got are of correct

   * sizes */

  insize = samples * this->in_info.bpf;

  outsize = samples * this->out_info.bpf;

  if (insize == 0 || outsize == 0)

    return GST_FLOW_OK;

  gst_buffer_resize (outbuf, 0, outsize);

  /* get src and dst data */

  if (inbuf != outbuf) {

    inbuf_writable = gst_buffer_is_writable (inbuf)

        && gst_buffer_n_memory (inbuf) == 1

        && gst_memory_is_writable (gst_buffer_peek_memory (inbuf, 0));

    if (!gst_buffer_map (inbuf, &srcmap,

            inbuf_writable ? GST_MAP_READWRITE : GST_MAP_READ))

      goto inmap_error;

  } else {

    inbuf_writable = TRUE;

  }

  if (!gst_buffer_map (outbuf, &dstmap, GST_MAP_WRITE))

    goto outmap_error;

  /* check in and outsize */

  if (inbuf != outbuf) {

    if (srcmap.size < insize)

      goto wrong_size;

  }

  if (dstmap.size < outsize)

    goto wrong_size;

  /* and convert the samples */

  flags = 0;

  if (inbuf_writable)

    flags |= GST_AUDIO_CONVERTER_FLAG_IN_WRITABLE;

  if (!GST_BUFFER_FLAG_IS_SET (inbuf, GST_BUFFER_FLAG_GAP)) {

    gpointer in[1] = { srcmap.data };

    gpointer out[1] = { dstmap.data };

    if (!gst_audio_converter_samples (this->convert, flags,

            inbuf != outbuf ? in : out, samples, out, samples))

      goto convert_error;

  } else {

    /* Create silence buffer */

    gst_audio_format_fill_silence (this->out_info.finfo, dstmap.data, outsize);

  }

  ret = GST_FLOW_OK;

done:

  gst_buffer_unmap (outbuf, &dstmap);

  if (inbuf != outbuf)

    gst_buffer_unmap (inbuf, &srcmap);

  return ret;

  /* ERRORS */

wrong_size:

  {

    GST_ELEMENT_ERROR (this, STREAM, FORMAT,

        (NULL),

        ("input/output buffers are of wrong size in: %" G_GSIZE_FORMAT " < %d"

            " or out: %" G_GSIZE_FORMAT " < %d",

            srcmap.size, insize, dstmap.size, outsize));

    ret = GST_FLOW_ERROR;

    goto done;

  }

convert_error:

  {

    GST_ELEMENT_ERROR (this, STREAM, FORMAT,

        (NULL), ("error while converting"));

    ret = GST_FLOW_ERROR;

    goto done;

  }

inmap_error: