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gst/audioscale/gstaudioscale.*: made audioscale resample from any sample rate to any sample rate
Original commit message from CVS: 2004-08-17 Zaheer Abbas Merali <zaheerabbas at merali dot org> * gst/audioscale/gstaudioscale.c: * gst/audioscale/gstaudioscale.h: made audioscale resample from any sample rate to any sample rate
This commit is contained in:
Zaheer Abbas Merali
parent
c718585b97
commit
d9e22cf818
3 changed files with 224 additions and 34 deletions
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@ -1,3 +1,9 @@
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2004-08-17 Zaheer Abbas Merali <zaheerabbas at merali dot org>
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* gst/audioscale/gstaudioscale.c:
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* gst/audioscale/gstaudioscale.h:
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made audioscale resample from any sample rate to any sample rate
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2004-08-17 Thomas Vander Stichele <thomas at apestaart dot org>
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* ext/libpng/gstpngdec.c:
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@ -30,6 +30,9 @@
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#include <gst/audio/audio.h>
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#include <gst/resample/resample.h>
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GST_DEBUG_CATEGORY_STATIC (audioscale_debug);
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#define GST_CAT_DEFAULT audioscale_debug
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/* elementfactory information */
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static GstElementDetails gst_audioscale_details =
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GST_ELEMENT_DETAILS ("Audio scaler",
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@ -112,6 +115,8 @@ static void gst_audioscale_set_property (GObject * object, guint prop_id,
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static void gst_audioscale_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec);
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static void *gst_audioscale_get_buffer (void *priv, unsigned int size);
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static GstElementClass *parent_class = NULL;
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/*static guint gst_audioscale_signals[LAST_SIGNAL] = { 0 }; */
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@ -173,6 +178,9 @@ static void gst_audioscale_class_init (AudioscaleClass * klass)
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G_PARAM_READWRITE | G_PARAM_CONSTRUCT));
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parent_class = g_type_class_ref (GST_TYPE_ELEMENT);
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GST_DEBUG_CATEGORY_INIT (audioscale_debug, "audioscale", 0,
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"audioscale element");
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}
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static void gst_audioscale_expand_value (GValue * dest, const GValue * src)
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@ -191,14 +199,8 @@ static void gst_audioscale_expand_value (GValue * dest, const GValue * src)
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rate_max = gst_value_get_int_range_max (src);
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}
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rate_min = (rate_min + 1) / 2;
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if (rate_min < 1)
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rate_min = 1;
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if (rate_max < G_MAXINT / 2) {
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rate_max *= 2;
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} else {
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rate_max = G_MAXINT;
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}
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g_value_init (dest, GST_TYPE_INT_RANGE);
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gst_value_set_int_range (dest, rate_min, rate_max);
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@ -262,6 +264,7 @@ static void gst_audioscale_expand_caps (GstCaps * caps)
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/* we do this hack, because the audioscale lib doesn't handle
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* rate conversions larger than a factor of 2 */
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/* UPDATE: allowed for n iterations so can handle any factor */
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for (i = 0; i < gst_caps_get_size (caps); i++) {
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GstStructure *structure = gst_caps_get_structure (caps, i);
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const GValue *value;
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@ -308,7 +311,7 @@ static GstCaps *gst_audioscale_fixate (GstPad * pad, const GstCaps * caps)
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GstStructure *structure;
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audioscale = GST_AUDIOSCALE (gst_pad_get_parent (pad));
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r = audioscale->gst_resample;
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r = &(audioscale->gst_resample_template);
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if (pad == audioscale->srcpad)
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{
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otherpad = audioscale->sinkpad;
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@ -337,6 +340,8 @@ static GstPadLinkReturn gst_audioscale_link (GstPad * pad, const GstCaps * caps)
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gst_resample_t *r;
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GstStructure *structure;
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double *rate, *otherrate;
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double temprate;
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int temp;
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gboolean ret;
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GstPadLinkReturn link_ret;
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@ -344,7 +349,7 @@ static GstPadLinkReturn gst_audioscale_link (GstPad * pad, const GstCaps * caps)
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GstCaps *copy;
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audioscale = GST_AUDIOSCALE (gst_pad_get_parent (pad));
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r = audioscale->gst_resample;
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r = &(audioscale->gst_resample_template);
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if (pad == audioscale->srcpad)
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{
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@ -362,12 +367,11 @@ static GstPadLinkReturn gst_audioscale_link (GstPad * pad, const GstCaps * caps)
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ret = gst_structure_get_int (structure, "rate", &temp);
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ret &= gst_structure_get_int (structure, "channels", &r->channels);
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g_return_val_if_fail (ret, GST_PAD_LINK_REFUSED);
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*rate = temp;
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*rate = (double) temp;
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copy = gst_caps_copy (caps);
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gst_audioscale_expand_caps (copy);
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link_ret = gst_pad_try_set_caps_nonfixed (otherpad, copy);
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if (GST_PAD_LINK_FAILED (link_ret))
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return link_ret;
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@ -376,7 +380,7 @@ static GstPadLinkReturn gst_audioscale_link (GstPad * pad, const GstCaps * caps)
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structure = gst_caps_get_structure (caps, 0);
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ret = gst_structure_get_int (structure, "rate", &temp);
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g_return_val_if_fail (ret, GST_PAD_LINK_REFUSED);
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*otherrate = temp;
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*otherrate = (double) temp;
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if (g_str_equal (gst_structure_get_name (structure), "audio/x-raw-float")) {
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r->format = GST_RESAMPLE_FLOAT;
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} else {
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@ -384,7 +388,67 @@ static GstPadLinkReturn gst_audioscale_link (GstPad * pad, const GstCaps * caps)
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}
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audioscale->passthru = (r->i_rate == r->o_rate);
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gst_resample_reinit (r);
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audioscale->increase = (r->o_rate >= r->i_rate);
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/* now create audioscale iterations */
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audioscale->num_iterations = 0;
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temprate = r->i_rate;
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while (TRUE) {
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if (r->o_rate > r->i_rate) {
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if (temprate >= r->o_rate)
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break;
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temprate *= 2;
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} else {
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if (temprate <= r->o_rate)
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break;
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temprate /= 2;
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}
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audioscale->num_iterations++;
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}
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if (audioscale->num_iterations > 0) {
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audioscale->offsets = g_new0 (gint64, audioscale->num_iterations);
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audioscale->gst_resample = g_new0 (gst_resample_t, 1);
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audioscale->gst_resample->priv = audioscale;
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audioscale->gst_resample->get_buffer = gst_audioscale_get_buffer;
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audioscale->gst_resample->method = r->method;
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audioscale->gst_resample->channels = r->channels;
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audioscale->gst_resample->filter_length = r->filter_length;
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audioscale->gst_resample->format = r->format;
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if (audioscale->increase) {
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temprate = r->o_rate;
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while (temprate / 2 >= r->i_rate) {
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temprate = temprate / 2;
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}
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/* now temprate is output rate of gstresample */
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GST_DEBUG ("gstresample will increase rate from %f to %f", r->i_rate,
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temprate);
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audioscale->gst_resample->o_rate = temprate;
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audioscale->gst_resample->i_rate = r->i_rate;
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} else {
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temprate = r->i_rate;
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while (temprate / 2 >= r->o_rate) {
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temprate = temprate / 2;
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}
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/* now temprate is input rate of gstresample */
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GST_DEBUG ("gstresample will decrease rate from %f to %f", temprate,
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r->o_rate);
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audioscale->gst_resample->o_rate = r->o_rate;
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audioscale->gst_resample->i_rate = temprate;
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}
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audioscale->passthru =
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(audioscale->gst_resample->i_rate == audioscale->gst_resample->o_rate);
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if (!audioscale->passthru)
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audioscale->num_iterations--;
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GST_DEBUG ("Number of iterations: %d", audioscale->num_iterations);
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gst_resample_init (audioscale->gst_resample);
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}
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return link_ret;
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}
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@ -393,21 +457,101 @@ static void *gst_audioscale_get_buffer (void *priv, unsigned int size)
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{
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Audioscale *audioscale = priv;
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GST_DEBUG ("size requested: %u irate: %f orate: %f", size,
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audioscale->gst_resample->i_rate, audioscale->gst_resample->o_rate);
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audioscale->outbuf = gst_buffer_new ();
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GST_BUFFER_SIZE (audioscale->outbuf) = size;
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GST_BUFFER_DATA (audioscale->outbuf) = g_malloc (size);
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GST_BUFFER_TIMESTAMP (audioscale->outbuf) =
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audioscale->offset * GST_SECOND / audioscale->gst_resample->o_rate;
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audioscale->offset +=
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audioscale->gst_resample_offset * GST_SECOND /
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audioscale->gst_resample->o_rate;
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audioscale->gst_resample_offset +=
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size / sizeof (gint16) / audioscale->gst_resample->channels;
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return GST_BUFFER_DATA (audioscale->outbuf);
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}
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/* reduces rate by factor of 2 */
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GstBuffer *gst_audioscale_decrease_rate (Audioscale * audioscale,
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GstBuffer * buf, double outrate, int cur_iteration)
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{
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gint i, j, curoffset;
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GstBuffer *outbuf = gst_buffer_new ();
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gint16 *outdata;
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gint16 *indata;
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GST_BUFFER_SIZE (outbuf) = GST_BUFFER_SIZE (buf) / 2;
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outdata = g_malloc (GST_BUFFER_SIZE (outbuf));
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indata = (gint16 *) GST_BUFFER_DATA (buf);
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GST_DEBUG
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("iteration = %d channels = %d in size = %d out size = %d outrate = %f",
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cur_iteration, audioscale->gst_resample_template.channels,
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GST_BUFFER_SIZE (buf), GST_BUFFER_SIZE (outbuf), outrate);
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curoffset = 0;
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for (i = 0; i < GST_BUFFER_SIZE (buf) / (sizeof (gint16));
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i += 2 * audioscale->gst_resample_template.channels) {
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for (j = 0; j < audioscale->gst_resample_template.channels; j++) {
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outdata[curoffset + j] =
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(indata[i + j] + indata[i + j +
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audioscale->gst_resample_template.channels]) / 2;
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}
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curoffset += audioscale->gst_resample_template.channels;
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}
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GST_BUFFER_DATA (outbuf) = (gpointer) outdata;
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GST_BUFFER_TIMESTAMP (outbuf) =
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audioscale->offsets[cur_iteration] * GST_SECOND / outrate;
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audioscale->offsets[cur_iteration] +=
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GST_BUFFER_SIZE (outbuf) / sizeof (gint16) /
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audioscale->gst_resample->channels;
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return outbuf;
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}
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/* increases rate by factor of 2 */
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GstBuffer *gst_audioscale_increase_rate (Audioscale * audioscale,
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GstBuffer * buf, double outrate, int cur_iteration)
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{
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gint i, j, curoffset;
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GstBuffer *outbuf = gst_buffer_new ();
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gint16 *outdata;
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gint16 *indata;
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GST_BUFFER_SIZE (outbuf) = GST_BUFFER_SIZE (buf) * 2;
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outdata = g_malloc (GST_BUFFER_SIZE (outbuf));
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indata = (gint16 *) GST_BUFFER_DATA (buf);
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GST_DEBUG
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("iteration = %d channels = %d in size = %d out size = %d out rate = %f",
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cur_iteration, audioscale->gst_resample_template.channels,
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GST_BUFFER_SIZE (buf), GST_BUFFER_SIZE (outbuf), outrate);
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curoffset = 0;
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for (i = 0; i < GST_BUFFER_SIZE (buf) / (sizeof (gint16));
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i += audioscale->gst_resample_template.channels) {
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for (j = 0; j < audioscale->gst_resample_template.channels; j++) {
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outdata[curoffset] = indata[i + j];
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outdata[curoffset + audioscale->gst_resample_template.channels] =
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indata[i + j];
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curoffset++;
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}
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curoffset += audioscale->gst_resample_template.channels;
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}
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GST_BUFFER_DATA (outbuf) = (gpointer) outdata;
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GST_BUFFER_TIMESTAMP (outbuf) =
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audioscale->offsets[cur_iteration] * GST_SECOND / outrate;
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audioscale->offsets[cur_iteration] +=
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GST_BUFFER_SIZE (outbuf) / sizeof (gint16) /
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audioscale->gst_resample->channels;
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return outbuf;
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}
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static void gst_audioscale_init (Audioscale * audioscale)
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{
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gst_resample_t *r;
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audioscale->num_iterations = 1;
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audioscale->sinkpad =
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gst_pad_new_from_template (gst_static_pad_template_get
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(&gst_audioscale_sink_template), "sink");
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@ -426,8 +570,7 @@ static void gst_audioscale_init (Audioscale * audioscale)
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gst_pad_set_getcaps_function (audioscale->srcpad, gst_audioscale_getcaps);
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gst_pad_set_fixate_function (audioscale->srcpad, gst_audioscale_fixate);
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r = g_new0 (gst_resample_t, 1);
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audioscale->gst_resample = r;
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r = &(audioscale->gst_resample_template);
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r->priv = audioscale;
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r->get_buffer = gst_audioscale_get_buffer;
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@ -439,35 +582,42 @@ static void gst_audioscale_init (Audioscale * audioscale)
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r->format = GST_RESAMPLE_S16;
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/*r->verbose = 1; */
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gst_resample_init (r);
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/* we will be reinitialized when the G_PARAM_CONSTRUCTs hit */
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audioscale->gst_resample = NULL;
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audioscale->outbuf = NULL;
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audioscale->offsets = NULL;
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audioscale->gst_resample_offset = 0;
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audioscale->increase = FALSE;
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}
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static void gst_audioscale_dispose (GObject * object)
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{
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Audioscale *audioscale = GST_AUDIOSCALE (object);
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if (audioscale->gst_resample)
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if (audioscale->gst_resample) {
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g_free (audioscale->gst_resample);
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audioscale->gst_resample = NULL;
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}
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if (audioscale->offsets)
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g_free (audioscale->offsets);
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G_OBJECT_CLASS (parent_class)->dispose (object);
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}
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static void gst_audioscale_chain (GstPad * pad, GstData * _data)
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{
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GstBuffer *buf = GST_BUFFER (_data);
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GstBuffer *tempbuf, *tempbuf2;
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Audioscale *audioscale;
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guchar *data;
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gulong size;
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gint i;
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double outrate;
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g_return_if_fail (pad != NULL);
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g_return_if_fail (GST_IS_PAD (pad));
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g_return_if_fail (buf != NULL);
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audioscale = GST_AUDIOSCALE (gst_pad_get_parent (pad));
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if (audioscale->passthru) {
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if (audioscale->passthru && audioscale->num_iterations == 0) {
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gst_pad_push (audioscale->srcpad, GST_DATA (buf));
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return;
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}
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@ -478,11 +628,41 @@ static void gst_audioscale_chain (GstPad * pad, GstData * _data)
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GST_DEBUG ("gst_audioscale_chain: got buffer of %ld bytes in '%s'\n",
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size, gst_element_get_name (GST_ELEMENT (audioscale)));
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tempbuf = buf;
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outrate = audioscale->gst_resample_template.i_rate;
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if (audioscale->increase && !audioscale->passthru) {
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GST_DEBUG ("doing gstresample");
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gst_resample_scale (audioscale->gst_resample, data, size);
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gst_pad_push (audioscale->srcpad, GST_DATA (audioscale->outbuf));
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tempbuf = audioscale->outbuf;
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gst_buffer_unref (buf);
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outrate = audioscale->gst_resample->o_rate;
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}
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for (i = 0; i < audioscale->num_iterations; i++) {
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tempbuf2 = tempbuf;
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GST_DEBUG ("doing %s",
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audioscale->
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increase ? "gst_audioscale_increase_rate" :
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"gst_audioscale_decrease_rate");
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if (audioscale->increase) {
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outrate *= 2;
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tempbuf = gst_audioscale_increase_rate (audioscale, tempbuf, outrate, i);
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} else {
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outrate /= 2;
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tempbuf = gst_audioscale_decrease_rate (audioscale, tempbuf, outrate, i);
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}
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gst_buffer_unref (tempbuf2);
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data = GST_BUFFER_DATA (tempbuf);
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size = GST_BUFFER_SIZE (tempbuf);
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}
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if (!audioscale->increase && !audioscale->passthru) {
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gst_resample_scale (audioscale->gst_resample, data, size);
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gst_buffer_unref (tempbuf);
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tempbuf = audioscale->outbuf;
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}
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gst_pad_push (audioscale->srcpad, GST_DATA (tempbuf));
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}
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static void
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@ -495,7 +675,7 @@ 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:
|
||||
|
|
@ -520,7 +700,7 @@ static void
|
|||
gst_resample_t *r;
|
||||
|
||||
src = GST_AUDIOSCALE (object);
|
||||
r = src->gst_resample;
|
||||
r = &(src->gst_resample_template);
|
||||
|
||||
switch (prop_id) {
|
||||
case ARG_FILTERLEN:
|
||||
|
|
|
|||
|
|
@ -57,10 +57,14 @@ struct _Audioscale {
|
|||
|
||||
/* audio state */
|
||||
gboolean passthru;
|
||||
gint64 offset;
|
||||
gint64 gst_resample_offset;
|
||||
|
||||
gint64* offsets;
|
||||
gboolean increase; /* is the rate change an increase */
|
||||
gint num_iterations; /* number of iterations through gst_audioscale/(increase|decrease)_rate */
|
||||
|
||||
gst_resample_t gst_resample_template;
|
||||
gst_resample_t* gst_resample;
|
||||
|
||||
GstBuffer* outbuf;
|
||||
};
|
||||
|
||||
|
|
|
|||
Loading…
Reference in a new issue