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gst/audioscale/gstaudioscale.c: %#^@^#@^@#^#@^#@^@#^@#^@#^@#^#@^#@^#@^@#^#@ fix seeking when resampling - how the ^@$...
Original commit message from CVS: * gst/audioscale/gstaudioscale.c: (gst_audioscale_init), (gst_audioscale_chain): %#^@^#@^@#^#@^#@^@#^@#^@#^@#^#@^#@^#@^@#^#@ fix seeking when resampling - how the ^@$^!@^! is this possible?!?
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@ -1,3 +1,10 @@
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2005-01-09 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
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* gst/audioscale/gstaudioscale.c: (gst_audioscale_init),
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(gst_audioscale_chain):
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%#^@^#@^@#^#@^#@^@#^@#^@#^@#^#@^#@^#@^@#^#@ fix seeking
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when resampling - how the ^@$^!@^! is this possible?!?
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2005-01-09 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
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* ext/alsa/gstalsa.c: (gst_alsa_change_state):
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@ -523,6 +523,8 @@ gst_audioscale_init (Audioscale * audioscale)
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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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GST_FLAG_SET (audioscale, GST_ELEMENT_EVENT_AWARE);
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}
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static void
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@ -564,6 +566,45 @@ gst_audioscale_chain (GstPad * pad, GstData * _data)
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return;
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}
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if (GST_IS_EVENT (_data)) {
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GstEvent *e = GST_EVENT (_data);
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switch (GST_EVENT_TYPE (e)) {
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case GST_EVENT_DISCONTINUOUS:{
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gint64 new_off;
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if (gst_event_discont_get_value (e, GST_FORMAT_TIME, &new_off)) {
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/* time -> out-sample */
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new_off = new_off * audioscale->gst_resample->o_rate / GST_SECOND;
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} else if (gst_event_discont_get_value (e,
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GST_FORMAT_DEFAULT, &new_off)) {
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/* in-sample -> out-sample */
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new_off *= audioscale->gst_resample->o_rate;
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new_off /= audioscale->gst_resample->i_rate;
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} else if (gst_event_discont_get_value (e, GST_FORMAT_BYTES, &new_off)) {
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new_off /= audioscale->gst_resample->channels;
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new_off /=
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(audioscale->gst_resample->format == GST_RESAMPLE_S16) ? 2 : 4;
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new_off *= audioscale->gst_resample->o_rate;
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new_off /= audioscale->gst_resample->i_rate;
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} else {
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/* *sigh* */
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new_off = 0;
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}
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audioscale->gst_resample_offset = new_off;
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/* fall-through */
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}
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default:
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gst_pad_event_default (pad, e);
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break;
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}
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return;
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} else if (GST_BUFFER_TIMESTAMP_IS_VALID (buf)) {
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/* update time for out-sample */
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audioscale->gst_resample_offset = GST_BUFFER_TIMESTAMP (buf) *
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audioscale->gst_resample->o_rate / GST_SECOND;
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}
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data = GST_BUFFER_DATA (buf);
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size = GST_BUFFER_SIZE (buf);
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