/* Element-Checklist-Version: 5 */

 * <refsect2>

 * Audioresample resamples raw audio buffers to different sample rates using

 * <para>

 * <programlisting>

 * </programlisting>

 * Decode an Ogg/Vorbis downsample to 8Khz and play sound through alsa. 

 * </para>

 *

 * Last reviewed on 2006-03-02 (0.10.4)

/*#define DEBUG_ENABLED */

GST_DEBUG_CATEGORY_STATIC (audioresample_debug);

#define GST_CAT_DEFAULT audioresample_debug

/* elementfactory information */

static const GstElementDetails gst_audioresample_details =

GST_ELEMENT_DETAILS ("Audio scaler",

    "Filter/Converter/Audio",

    "Resample audio",

    "David Schleef <ds@schleef.org>");

#define DEFAULT_FILTERLEN       16

  PROP_FILTERLEN

      "signed = (boolean) true;" \

      "width = (int) 32, " \

      "depth = (int) 32, " \

      "signed = (boolean) true;" \

    "audio/x-raw-float, " \

      "rate = (int) [ 1, MAX ], "	\

      "channels = (int) [ 1, MAX ], " \

      "endianness = (int) BYTE_ORDER, " \

      "width = (int) 32; " \

    "audio/x-raw-float, " \

      "rate = (int) [ 1, MAX ], "	\

      "channels = (int) [ 1, MAX ], " \

      "endianness = (int) BYTE_ORDER, " \

      "width = (int) 64" \

static GstStaticPadTemplate gst_audioresample_sink_template =

static GstStaticPadTemplate gst_audioresample_src_template =

static void gst_audioresample_set_property (GObject * object,

static void gst_audioresample_get_property (GObject * object,

static gboolean audioresample_get_unit_size (GstBaseTransform * base,

static GstCaps *audioresample_transform_caps (GstBaseTransform * base,

static gboolean audioresample_transform_size (GstBaseTransform * trans,

static gboolean audioresample_set_caps (GstBaseTransform * base,

static GstFlowReturn audioresample_pushthrough (GstAudioresample *

    audioresample);

static GstFlowReturn audioresample_transform (GstBaseTransform * base,

static gboolean audioresample_event (GstBaseTransform * base, GstEvent * event);

static gboolean audioresample_start (GstBaseTransform * base);

static gboolean audioresample_stop (GstBaseTransform * base);

static gboolean audioresample_query (GstPad * pad, GstQuery * query);

static const GstQueryType *audioresample_query_type (GstPad * pad);

#define DEBUG_INIT(bla) \

  GST_DEBUG_CATEGORY_INIT (audioresample_debug, "audioresample", 0, "audio resampling element");

GST_BOILERPLATE_FULL (GstAudioresample, gst_audioresample, GstBaseTransform,

    GST_TYPE_BASE_TRANSFORM, DEBUG_INIT);

gst_audioresample_base_init (gpointer g_class)

      gst_static_pad_template_get (&gst_audioresample_src_template));

      gst_static_pad_template_get (&gst_audioresample_sink_template));

  gst_element_class_set_details (gstelement_class, &gst_audioresample_details);

gst_audioresample_class_init (GstAudioresampleClass * klass)

  GObjectClass *gobject_class;

  gobject_class = (GObjectClass *) klass;

  gobject_class->set_property = gst_audioresample_set_property;

  gobject_class->get_property = gst_audioresample_get_property;

  g_object_class_install_property (gobject_class, PROP_FILTERLEN,

      g_param_spec_int ("filter_length", "filter_length", "filter_length",

          0, G_MAXINT, DEFAULT_FILTERLEN,

      GST_DEBUG_FUNCPTR (audioresample_start);

      GST_DEBUG_FUNCPTR (audioresample_stop);

      GST_DEBUG_FUNCPTR (audioresample_transform_size);

      GST_DEBUG_FUNCPTR (audioresample_get_unit_size);

      GST_DEBUG_FUNCPTR (audioresample_transform_caps);

      GST_DEBUG_FUNCPTR (audioresample_set_caps);

      GST_DEBUG_FUNCPTR (audioresample_transform);

      GST_DEBUG_FUNCPTR (audioresample_event);

gst_audioresample_init (GstAudioresample * audioresample,

    GstAudioresampleClass * klass)

  GstBaseTransform *trans;

  trans = GST_BASE_TRANSFORM (audioresample);

  /* 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;

  audioresample->need_discont = FALSE;

  gst_pad_set_query_function (trans->srcpad, audioresample_query);

  gst_pad_set_query_type_function (trans->srcpad, audioresample_query_type);

audioresample_start (GstBaseTransform * base)

  GstAudioresample *audioresample = GST_AUDIORESAMPLE (base);

  audioresample->resample = resample_new ();

  audioresample->ts_offset = -1;

  audioresample->offset = -1;

  audioresample->next_ts = -1;

  resample_set_filter_length (audioresample->resample,

      audioresample->filter_length);

audioresample_stop (GstBaseTransform * base)

  GstAudioresample *audioresample = GST_AUDIORESAMPLE (base);

  if (audioresample->resample) {

    resample_free (audioresample->resample);

    audioresample->resample = NULL;

  gst_caps_replace (&audioresample->sinkcaps, NULL);

  gst_caps_replace (&audioresample->srccaps, NULL);

audioresample_get_unit_size (GstBaseTransform * base, GstCaps * caps,

  g_assert (size);

  g_return_val_if_fail (ret, FALSE);

  *size = width * channels / 8;

audioresample_transform_caps (GstBaseTransform * base,

  GstStructure *structure;

  /* transform caps gives one single caps so we can just replace

   * the rate property with our range. */

  res = gst_caps_copy (caps);

  structure = gst_caps_get_structure (res, 0);

  gst_structure_set (structure, "rate", GST_TYPE_INT_RANGE, 1, G_MAXINT, NULL);

resample_set_state_from_caps (ResampleState * state, GstCaps * incaps,

    GstCaps * outcaps, gint * channels, gint * inrate, gint * outrate)

  gint myinrate, myoutrate;

  int mychannels;

  gint width, depth;

  ResampleFormat format;

  /* get width */

  ret = gst_structure_get_int (structure, "width", &width);

  if (!ret)

    goto no_width;

  /* figure out the format */

  if (g_str_equal (gst_structure_get_name (structure), "audio/x-raw-float")) {

    if (width == 32)

      format = RESAMPLE_FORMAT_F32;

    else if (width == 64)

      format = RESAMPLE_FORMAT_F64;

    else

      goto wrong_depth;

  } else {

    /* for int, depth and width must be the same */

    ret = gst_structure_get_int (structure, "depth", &depth);

    if (!ret || width != depth)

      goto not_equal;

    if (width == 16)

      format = RESAMPLE_FORMAT_S16;

    else if (width == 32)

      format = RESAMPLE_FORMAT_S32;

    else

      goto wrong_depth;

  }

  if (!ret)

  if (!ret)

  resample_set_format (state, format);

  resample_set_n_channels (state, mychannels);

  resample_set_input_rate (state, myinrate);

  resample_set_output_rate (state, myoutrate);

no_width:

  {

    GST_DEBUG ("failed to get width from caps");

    return FALSE;

  }

not_equal:

  {

    GST_DEBUG ("width %d and depth %d must be the same", width, depth);

    return FALSE;

  }

wrong_depth:

  {

    GST_DEBUG ("unknown depth %d found", depth);

    return FALSE;

  }

audioresample_transform_size (GstBaseTransform * base,

  GstAudioresample *audioresample = GST_AUDIORESAMPLE (base);

  ResampleState *state;

  gboolean use_internal = FALSE;        /* whether we use the internal state */

  /* if the caps are the ones that _set_caps got called with; we can use

   * our own state; otherwise we'll have to create a state */

  if (gst_caps_is_equal (sinkcaps, audioresample->sinkcaps) &&

      gst_caps_is_equal (srccaps, audioresample->srccaps)) {

    use_internal = TRUE;

    state = audioresample->resample;

  } else {

    GST_DEBUG_OBJECT (audioresample,

        "caps are not the set caps, creating state");

    state = resample_new ();

    resample_set_filter_length (state, audioresample->filter_length);

    resample_set_state_from_caps (state, sinkcaps, srccaps, NULL, NULL, NULL);

    /* asked to convert size of an incoming buffer */

    *othersize = resample_get_output_size_for_input (state, size);

    /* asked to convert size of an outgoing buffer */

    *othersize = resample_get_input_size_for_output (state, size);

  g_assert (*othersize % state->sample_size == 0);

  /* we make room for one extra sample, given that the resampling filter

   * can output an extra one for non-integral i_rate/o_rate */

  GST_LOG_OBJECT (base, "transformed size %d to %d", size, *othersize);

  if (!use_internal) {

    resample_free (state);

  }

audioresample_set_caps (GstBaseTransform * base, GstCaps * incaps,

  gint inrate, outrate;

  int channels;

  GstAudioresample *audioresample = GST_AUDIORESAMPLE (base);

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

  ret = resample_set_state_from_caps (audioresample->resample, incaps, outcaps,

      &channels, &inrate, &outrate);

  g_return_val_if_fail (ret, FALSE);

  audioresample->channels = channels;

  GST_DEBUG_OBJECT (audioresample, "set channels to %d", channels);

  audioresample->i_rate = inrate;

  GST_DEBUG_OBJECT (audioresample, "set i_rate to %d", inrate);

  audioresample->o_rate = outrate;

  GST_DEBUG_OBJECT (audioresample, "set o_rate to %d", outrate);

  gst_caps_replace (&audioresample->sinkcaps, incaps);

  gst_caps_replace (&audioresample->srccaps, outcaps);

  return TRUE;

}

static gboolean

audioresample_event (GstBaseTransform * base, GstEvent * event)

{

  GstAudioresample *audioresample;

  audioresample = GST_AUDIORESAMPLE (base);

  switch (GST_EVENT_TYPE (event)) {

    case GST_EVENT_FLUSH_START:

      break;

    case GST_EVENT_FLUSH_STOP:

      resample_input_flush (audioresample->resample);

      audioresample->ts_offset = -1;

      audioresample->next_ts = -1;

      audioresample->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;

      break;

    case GST_EVENT_EOS:

      resample_input_eos (audioresample->resample);

      audioresample_pushthrough (audioresample);

      break;

    default:

      break;

  }

  parent_class->event (base, event);

static GstFlowReturn

audioresample_do_output (GstAudioresample * audioresample, GstBuffer * outbuf)

  int outsize;

  int outsamples;

  ResampleState *r;