#include <gst/audio/audio.h>
#include <gst/resample/resample.h>
+GST_DEBUG_CATEGORY_STATIC (audioscale_debug);
+#define GST_CAT_DEFAULT audioscale_debug
+
/* elementfactory information */
static GstElementDetails gst_audioscale_details =
GST_ELEMENT_DETAILS ("Audio scaler",
static void gst_audioscale_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
+static void *gst_audioscale_get_buffer (void *priv, unsigned int size);
+
static GstElementClass *parent_class = NULL;
/*static guint gst_audioscale_signals[LAST_SIGNAL] = { 0 }; */
G_PARAM_READWRITE | G_PARAM_CONSTRUCT));
parent_class = g_type_class_ref (GST_TYPE_ELEMENT);
+
+ GST_DEBUG_CATEGORY_INIT (audioscale_debug, "audioscale", 0,
+ "audioscale element");
}
static void gst_audioscale_expand_value (GValue * dest, const GValue * src)
rate_max = gst_value_get_int_range_max (src);
}
- rate_min = (rate_min + 1) / 2;
- if (rate_min < 1)
- rate_min = 1;
- if (rate_max < G_MAXINT / 2) {
- rate_max *= 2;
- } else {
- rate_max = G_MAXINT;
- }
+ rate_min = 1;
+ rate_max = G_MAXINT;
g_value_init (dest, GST_TYPE_INT_RANGE);
gst_value_set_int_range (dest, rate_min, rate_max);
/* we do this hack, because the audioscale lib doesn't handle
* rate conversions larger than a factor of 2 */
+ /* UPDATE: allowed for n iterations so can handle any factor */
for (i = 0; i < gst_caps_get_size (caps); i++) {
GstStructure *structure = gst_caps_get_structure (caps, i);
const GValue *value;
GstStructure *structure;
audioscale = GST_AUDIOSCALE (gst_pad_get_parent (pad));
- r = audioscale->gst_resample;
+ r = &(audioscale->gst_resample_template);
if (pad == audioscale->srcpad)
{
otherpad = audioscale->sinkpad;
gst_resample_t *r;
GstStructure *structure;
double *rate, *otherrate;
+ double temprate;
+
int temp;
gboolean ret;
GstPadLinkReturn link_ret;
GstCaps *copy;
audioscale = GST_AUDIOSCALE (gst_pad_get_parent (pad));
- r = audioscale->gst_resample;
+ r = &(audioscale->gst_resample_template);
if (pad == audioscale->srcpad)
{
ret = gst_structure_get_int (structure, "rate", &temp);
ret &= gst_structure_get_int (structure, "channels", &r->channels);
g_return_val_if_fail (ret, GST_PAD_LINK_REFUSED);
- *rate = temp;
+ *rate = (double) temp;
copy = gst_caps_copy (caps);
gst_audioscale_expand_caps (copy);
link_ret = gst_pad_try_set_caps_nonfixed (otherpad, copy);
-
if (GST_PAD_LINK_FAILED (link_ret))
return link_ret;
structure = gst_caps_get_structure (caps, 0);
ret = gst_structure_get_int (structure, "rate", &temp);
g_return_val_if_fail (ret, GST_PAD_LINK_REFUSED);
- *otherrate = temp;
+ *otherrate = (double) temp;
if (g_str_equal (gst_structure_get_name (structure), "audio/x-raw-float")) {
r->format = GST_RESAMPLE_FLOAT;
} else {
}
audioscale->passthru = (r->i_rate == r->o_rate);
- gst_resample_reinit (r);
+ audioscale->increase = (r->o_rate >= r->i_rate);
+ /* now create audioscale iterations */
+ audioscale->num_iterations = 0;
+
+ temprate = r->i_rate;
+ while (TRUE) {
+ if (r->o_rate > r->i_rate) {
+ if (temprate >= r->o_rate)
+ break;
+ temprate *= 2;
+ } else {
+ if (temprate <= r->o_rate)
+ break;
+ temprate /= 2;
+ }
+ audioscale->num_iterations++;
+ }
+
+
+
+
+ if (audioscale->num_iterations > 0) {
+ audioscale->offsets = g_new0 (gint64, audioscale->num_iterations);
+ audioscale->gst_resample = g_new0 (gst_resample_t, 1);
+ audioscale->gst_resample->priv = audioscale;
+ audioscale->gst_resample->get_buffer = gst_audioscale_get_buffer;
+ audioscale->gst_resample->method = r->method;
+ audioscale->gst_resample->channels = r->channels;
+ audioscale->gst_resample->filter_length = r->filter_length;
+ audioscale->gst_resample->format = r->format;
+ if (audioscale->increase) {
+ temprate = r->o_rate;
+
+ while (temprate / 2 >= r->i_rate) {
+ temprate = temprate / 2;
+ }
+ /* now temprate is output rate of gstresample */
+ GST_DEBUG ("gstresample will increase rate from %f to %f", r->i_rate,
+ temprate);
+ audioscale->gst_resample->o_rate = temprate;
+ audioscale->gst_resample->i_rate = r->i_rate;
+ } else {
+ temprate = r->i_rate;
+
+ while (temprate / 2 >= r->o_rate) {
+ temprate = temprate / 2;
+ }
+ /* now temprate is input rate of gstresample */
+ GST_DEBUG ("gstresample will decrease rate from %f to %f", temprate,
+ r->o_rate);
+ audioscale->gst_resample->o_rate = r->o_rate;
+ audioscale->gst_resample->i_rate = temprate;
+ }
+ audioscale->passthru =
+ (audioscale->gst_resample->i_rate == audioscale->gst_resample->o_rate);
+ if (!audioscale->passthru)
+ audioscale->num_iterations--;
+ GST_DEBUG ("Number of iterations: %d", audioscale->num_iterations);
+
+ gst_resample_init (audioscale->gst_resample);
+ }
return link_ret;
}
{
Audioscale *audioscale = priv;
+ GST_DEBUG ("size requested: %u irate: %f orate: %f", size,
+ audioscale->gst_resample->i_rate, audioscale->gst_resample->o_rate);
audioscale->outbuf = gst_buffer_new ();
GST_BUFFER_SIZE (audioscale->outbuf) = size;
GST_BUFFER_DATA (audioscale->outbuf) = g_malloc (size);
GST_BUFFER_TIMESTAMP (audioscale->outbuf) =
- audioscale->offset * GST_SECOND / audioscale->gst_resample->o_rate;
- audioscale->offset +=
+ audioscale->gst_resample_offset * GST_SECOND /
+ audioscale->gst_resample->o_rate;
+ audioscale->gst_resample_offset +=
size / sizeof (gint16) / audioscale->gst_resample->channels;
return GST_BUFFER_DATA (audioscale->outbuf);
}
+/* reduces rate by factor of 2 */
+GstBuffer *gst_audioscale_decrease_rate (Audioscale * audioscale,
+ GstBuffer * buf, double outrate, int cur_iteration)
+{
+ gint i, j, curoffset;
+ GstBuffer *outbuf = gst_buffer_new ();
+ gint16 *outdata;
+ gint16 *indata;
+
+ GST_BUFFER_SIZE (outbuf) = GST_BUFFER_SIZE (buf) / 2;
+ outdata = g_malloc (GST_BUFFER_SIZE (outbuf));
+ indata = (gint16 *) GST_BUFFER_DATA (buf);
+
+ GST_DEBUG
+ ("iteration = %d channels = %d in size = %d out size = %d outrate = %f",
+ cur_iteration, audioscale->gst_resample_template.channels,
+ GST_BUFFER_SIZE (buf), GST_BUFFER_SIZE (outbuf), outrate);
+ curoffset = 0;
+ for (i = 0; i < GST_BUFFER_SIZE (buf) / (sizeof (gint16));
+ i += 2 * audioscale->gst_resample_template.channels) {
+ for (j = 0; j < audioscale->gst_resample_template.channels; j++) {
+ outdata[curoffset + j] =
+ (indata[i + j] + indata[i + j +
+ audioscale->gst_resample_template.channels]) / 2;
+ }
+ curoffset += audioscale->gst_resample_template.channels;
+ }
+
+ GST_BUFFER_DATA (outbuf) = (gpointer) outdata;
+ GST_BUFFER_TIMESTAMP (outbuf) =
+ audioscale->offsets[cur_iteration] * GST_SECOND / outrate;
+ audioscale->offsets[cur_iteration] +=
+ GST_BUFFER_SIZE (outbuf) / sizeof (gint16) /
+ audioscale->gst_resample->channels;
+ return outbuf;
+}
+
+/* increases rate by factor of 2 */
+GstBuffer *gst_audioscale_increase_rate (Audioscale * audioscale,
+ GstBuffer * buf, double outrate, int cur_iteration)
+{
+ gint i, j, curoffset;
+ GstBuffer *outbuf = gst_buffer_new ();
+ gint16 *outdata;
+ gint16 *indata;
+
+ GST_BUFFER_SIZE (outbuf) = GST_BUFFER_SIZE (buf) * 2;
+ outdata = g_malloc (GST_BUFFER_SIZE (outbuf));
+ indata = (gint16 *) GST_BUFFER_DATA (buf);
+
+ GST_DEBUG
+ ("iteration = %d channels = %d in size = %d out size = %d out rate = %f",
+ cur_iteration, audioscale->gst_resample_template.channels,
+ GST_BUFFER_SIZE (buf), GST_BUFFER_SIZE (outbuf), outrate);
+ curoffset = 0;
+ for (i = 0; i < GST_BUFFER_SIZE (buf) / (sizeof (gint16));
+ i += audioscale->gst_resample_template.channels) {
+ for (j = 0; j < audioscale->gst_resample_template.channels; j++) {
+ outdata[curoffset] = indata[i + j];
+ outdata[curoffset + audioscale->gst_resample_template.channels] =
+ indata[i + j];
+ curoffset++;
+ }
+ curoffset += audioscale->gst_resample_template.channels;
+ }
+
+ GST_BUFFER_DATA (outbuf) = (gpointer) outdata;
+ GST_BUFFER_TIMESTAMP (outbuf) =
+ audioscale->offsets[cur_iteration] * GST_SECOND / outrate;
+ audioscale->offsets[cur_iteration] +=
+ GST_BUFFER_SIZE (outbuf) / sizeof (gint16) /
+ audioscale->gst_resample->channels;
+ return outbuf;
+}
+
static void gst_audioscale_init (Audioscale * audioscale)
{
gst_resample_t *r;
+ audioscale->num_iterations = 1;
+
audioscale->sinkpad =
gst_pad_new_from_template (gst_static_pad_template_get
(&gst_audioscale_sink_template), "sink");
gst_pad_set_getcaps_function (audioscale->srcpad, gst_audioscale_getcaps);
gst_pad_set_fixate_function (audioscale->srcpad, gst_audioscale_fixate);
- r = g_new0 (gst_resample_t, 1);
- audioscale->gst_resample = r;
+ r = &(audioscale->gst_resample_template);
r->priv = audioscale;
r->get_buffer = gst_audioscale_get_buffer;
r->format = GST_RESAMPLE_S16;
/*r->verbose = 1; */
- gst_resample_init (r);
-
- /* we will be reinitialized when the G_PARAM_CONSTRUCTs hit */
+ audioscale->gst_resample = NULL;
+ audioscale->outbuf = NULL;
+ audioscale->offsets = NULL;
+ audioscale->gst_resample_offset = 0;
+ audioscale->increase = FALSE;
}
static void gst_audioscale_dispose (GObject * object)
{
Audioscale *audioscale = GST_AUDIOSCALE (object);
- if (audioscale->gst_resample)
+ if (audioscale->gst_resample) {
g_free (audioscale->gst_resample);
- audioscale->gst_resample = NULL;
-
+ }
+ if (audioscale->offsets)
+ g_free (audioscale->offsets);
G_OBJECT_CLASS (parent_class)->dispose (object);
}
static void gst_audioscale_chain (GstPad * pad, GstData * _data)
{
GstBuffer *buf = GST_BUFFER (_data);
+ GstBuffer *tempbuf, *tempbuf2;
+
Audioscale *audioscale;
guchar *data;
gulong size;
+ gint i;
+ double outrate;
g_return_if_fail (pad != NULL);
g_return_if_fail (GST_IS_PAD (pad));
g_return_if_fail (buf != NULL);
audioscale = GST_AUDIOSCALE (gst_pad_get_parent (pad));
- if (audioscale->passthru) {
+ if (audioscale->passthru && audioscale->num_iterations == 0) {
gst_pad_push (audioscale->srcpad, GST_DATA (buf));
return;
}
GST_DEBUG ("gst_audioscale_chain: got buffer of %ld bytes in '%s'\n",
size, gst_element_get_name (GST_ELEMENT (audioscale)));
- gst_resample_scale (audioscale->gst_resample, data, size);
+ tempbuf = buf;
+ outrate = audioscale->gst_resample_template.i_rate;
+ if (audioscale->increase && !audioscale->passthru) {
+ GST_DEBUG ("doing gstresample");
+ gst_resample_scale (audioscale->gst_resample, data, size);
+ tempbuf = audioscale->outbuf;
+ gst_buffer_unref (buf);
+ outrate = audioscale->gst_resample->o_rate;
+ }
+ for (i = 0; i < audioscale->num_iterations; i++) {
+ tempbuf2 = tempbuf;
+ GST_DEBUG ("doing %s",
+ audioscale->
+ increase ? "gst_audioscale_increase_rate" :
+ "gst_audioscale_decrease_rate");
+
+ if (audioscale->increase) {
+ outrate *= 2;
+ tempbuf = gst_audioscale_increase_rate (audioscale, tempbuf, outrate, i);
+ } else {
+ outrate /= 2;
+ tempbuf = gst_audioscale_decrease_rate (audioscale, tempbuf, outrate, i);
+ }
- gst_pad_push (audioscale->srcpad, GST_DATA (audioscale->outbuf));
+ gst_buffer_unref (tempbuf2);
+ data = GST_BUFFER_DATA (tempbuf);
+ size = GST_BUFFER_SIZE (tempbuf);
+ }
+ if (!audioscale->increase && !audioscale->passthru) {
+ gst_resample_scale (audioscale->gst_resample, data, size);
+ gst_buffer_unref (tempbuf);
+ tempbuf = audioscale->outbuf;
+ }
+ gst_pad_push (audioscale->srcpad, GST_DATA (tempbuf));
- gst_buffer_unref (buf);
}
static void
/* it's not null if we got it, but it might not be ours */
g_return_if_fail (GST_IS_AUDIOSCALE (object));
src = GST_AUDIOSCALE (object);
- r = src->gst_resample;
+ r = &(src->gst_resample_template);
switch (prop_id) {
case ARG_FILTERLEN:
gst_resample_t *r;
src = GST_AUDIOSCALE (object);
- r = src->gst_resample;
+ r = &(src->gst_resample_template);
switch (prop_id) {
case ARG_FILTERLEN:
projects / platform / upstream / gst-plugins-base.git / commitdiff