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adding TIMESTAMP to buffer ( work with goom now ;) adding AUTHORS code cleanup

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Original commit message from CVS:
adding TIMESTAMP to buffer ( work with goom now ;)
adding AUTHORS
code cleanup
This commit is contained in:
Jeremy Simon 2002-03-01 17:25:42 +00:00
parent add0caca4d
commit 870168f3d5
1 changed files with 25 additions and 10 deletions
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35
gst/modplug/gstmodplug.cc
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@ -16,6 +16,14 @@
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
/*
Code based on modplugxmms
XMMS plugin:
Kenton Varda <temporal@gauge3d.org>
Sound Engine:
Olivier Lapicque <olivierl@jps.net>
*/
#include "libmodplug/stdafx.h"
#include "libmodplug/sndfile.h"
@ -28,9 +36,11 @@
GstElementDetails modplug_details = {
"ModPlug",
"Audio/Module",
"Module decoder based on modplug engine from (Olivier ..)",
"Module decoder based on modplug engine",
VERSION,
"Jeremy SIMON <jsimon13@yahoo.fr>",
"Jeremy SIMON <jsimon13@yahoo.fr> "
"Kenton Varda <temporal@gauge3d.org> "
"Olivier Lapicque <olivierl@jps.net>",
"(C) 2001"
};
@ -260,7 +270,7 @@ gst_modplug_setup (GstModPlug *modplug)
modplug->mSoundFile->SetWaveConfig ( modplug->frequency, 8, modplug->channel );
modplug->mSoundFile->SetWaveConfigEx ( modplug->surround, !modplug->oversamp, modplug->reverb, true, modplug->megabass, modplug->noise_reduction, false );
modplug->mSoundFile->SetResamplingMode ( SRCMODE_SPLINE );
modplug->mSoundFile->SetResamplingMode ( SRCMODE_POLYPHASE );
if ( modplug->surround )
modplug->mSoundFile->SetSurroundParameters( modplug->surround_depth, modplug->surround_delay );
@ -281,6 +291,7 @@ gst_modplug_loop (GstElement *element)
GstModPlug *modplug;
GstBuffer *buffer_in, *buffer_out;
gint mode16bits;
guint64 total_samples, sync_point;
g_return_if_fail (element != NULL);
g_return_if_fail (GST_IS_MODPLUG (element));
@ -334,14 +345,14 @@ gst_modplug_loop (GstElement *element)
"rate", GST_PROPS_INT (modplug->frequency),
"channels", GST_PROPS_INT (modplug->channel)));
modplug->length = 512000 / modplug->frequency + 1;
modplug->length *= modplug->frequency;
modplug->length /= 1000;
modplug->length *= modplug->channel;
modplug->length = 1152 * modplug->channel;
if ( modplug->_16bit )
modplug->length *= 2;
modplug->audiobuffer = (guchar *) g_malloc( modplug->length );
total_samples = 0;
sync_point = 0;
do {
if( modplug->mSoundFile->Read ( modplug->audiobuffer, modplug->length ) != 0 )
@ -349,14 +360,18 @@ gst_modplug_loop (GstElement *element)
buffer_out = gst_buffer_new();
GST_BUFFER_DATA( buffer_out ) = (guchar *) g_memdup( modplug->audiobuffer, modplug->length );
GST_BUFFER_SIZE( buffer_out ) = modplug->length;
gst_pad_push( srcpad, buffer_out );
total_samples+=1152;
GST_BUFFER_TIMESTAMP( buffer_out ) = total_samples * 1000000LL / modplug->frequency;
gst_pad_push( srcpad, buffer_out );
gst_element_yield (element);
}
else
{
free( modplug->audiobuffer );
gst_element_set_eos (GST_ELEMENT (modplug));
gst_pad_push (modplug->srcpad, GST_BUFFER (gst_event_new (GST_EVENT_EOS)));
gst_pad_push (modplug->srcpad, GST_BUFFER (gst_event_new (GST_EVENT_EOS)));
}
}
while ( 1 );