commited by ljp (llornkcor) reverted to qpe's interface for compatibility. Added playlist features:

save, remove, and have different ones. Uncommented code for background image and added opielogo.

1 files changed, 9 insertions, 7 deletions

diff --git a/core/multimedia/opieplayer/wavplugin/wavplugin.cpp b/core/multimedia/opieplayer/wavplugin/wavplugin.cpp
index 7ac63c0..4a0da16 100644
--- a/core/multimedia/opieplayer/wavplugin/wavplugin.cpp
+++ b/core/multimedia/opieplayer/wavplugin/wavplugin.cpp
@@ -25,114 +25,116 @@
 #include <errno.h>
 #include <unistd.h>
 #include <qfile.h>
 #include "wavplugin.h"
 
 //#define debugMsg(a)     qDebug(a)
 #define debugMsg(a)
 
 
 struct RiffChunk {
     char id[4];
     Q_UINT32 size;
     char data[4];
 };
 
 
 struct ChunkData {
     Q_INT16 formatTag;
     Q_INT16 channels;
     Q_INT32 samplesPerSec;
     Q_INT32 avgBytesPerSec;
     Q_INT16 blockAlign;
     Q_INT16 wBitsPerSample;
 };
 
 
 const int sound_buffer_size = 512; // 4096; // you got to be kidding right?
 
 
 class WavPluginData {
 public:
     QFile *input;
 
     int wavedata_remaining;
     ChunkData chunkdata;
     RiffChunk chunk;
     uchar data[sound_buffer_size+32]; // +32 to handle badly aligned input data
     int out,max;
     int samples_due;
     int samples;
 
     WavPluginData() {
   max = out = sound_buffer_size;
   wavedata_remaining = 0;
   samples_due = 0;
   samples = -1;
     }
 
-    // expands out samples to the frequency of 44kHz
+    // expands out samples to the frequency of 44kHz //not any more
     bool add( short *output, long count, long& done, bool stereo )
     {
   done = 0;
     qApp->processEvents();
 
   if ( input == 0 ) {
       qDebug("no input");
       return FALSE;
   }
 
         while ( count ) {
             int l,r;
             if ( getSample(l, r) == FALSE ) {
                 qDebug("didn't get sample");
                 return FALSE;
             }
               samples_due += chunkdata.samplesPerSec;
+              printf("samples due %d\r", samples_due);
+              fflush(stdout);
              while ( count && (samples_due > chunkdata.samplesPerSec) ) {
                 *output++ = l;
                 if ( stereo )
                     *output++ = r;
                 samples_due -= chunkdata.samplesPerSec;
                 count--;
                 done++;
             }
         }
         return TRUE;
     }
 
     bool initialise() {
   if ( input == 0 )
       return FALSE;
 
   wavedata_remaining = -1;
 
   while ( wavedata_remaining == -1 ) {
       // Keep reading chunks...
       const int n = sizeof(chunk) - sizeof(chunk.data);
       int t = input->readBlock( (char*)&chunk, n );
       if ( t != n ) {
     if ( t == -1 )
         return FALSE;
     return TRUE;
       }
       if ( qstrncmp(chunk.id,"data",4) == 0 ) {
     samples = wavedata_remaining = chunk.size;
       } else if ( qstrncmp(chunk.id,"RIFF",4) == 0 ) {
     char d[4];
     if ( input->readBlock(d,4) != 4 ) {
         return FALSE;
     }
     if ( qstrncmp(d,"WAVE",4) != 0 ) {
         // skip
         if ( chunk.size > 1000000000 || !input->at(input->at()+chunk.size-4) ) {
       return FALSE;
         }
     }
       } else if ( qstrncmp(chunk.id,"fmt ",4) == 0 ) {
     if ( input->readBlock((char*)&chunkdata,sizeof(chunkdata)) != sizeof(chunkdata) ) {
         return FALSE;
     }
 #define WAVE_FORMAT_PCM 1
     if ( chunkdata.formatTag != WAVE_FORMAT_PCM ) {
         qDebug("WAV file: UNSUPPORTED FORMAT %d",chunkdata.formatTag);
         return FALSE;
@@ -283,58 +285,58 @@ int WavPlugin::audioSamples( int ) {
 //    qDebug( "WavPlugin::audioSamples" );
     return d->samples / d->chunkdata.channels/2; // ### Scaled samples will be made stereo,
         // Therefore if source is mono we will double the number of samples
 }
 
 
 bool WavPlugin::audioSetSample( long, int ) {
 //    qDebug( "WavPlugin::audioSetSample" );
     return FALSE;
 }
 
 
 long WavPlugin::audioGetSample( int ) {
 //    qDebug( "WavPlugin::audioGetSample" );
     return 0;
 }
 
 /*
 bool WavPlugin::audioReadSamples( short *, int, long, int ) {
     debugMsg( "WavPlugin::audioReadSamples" );
     return FALSE;
 }
 
 
 bool WavPlugin::audioReReadSamples( short *, int, long, int ) {
     debugMsg( "WavPlugin::audioReReadSamples" );
     return FALSE;
 }
 
 
 bool WavPlugin::audioReadMonoSamples( short *output, long samples, long& samplesMade, int ) {
     debugMsg( "WavPlugin::audioReadMonoSamples" );
     return !d->add( output, samples, samplesMade, FALSE );
 }
 
 
 bool WavPlugin::audioReadStereoSamples( short *output, long samples, long& samplesMade, int ) {
     debugMsg( "WavPlugin::audioReadStereoSamples" );
     return !d->add( output, samples, samplesMade, TRUE );
 }
 */
 
 bool WavPlugin::audioReadSamples( short *output, int channels, long samples, long& samplesMade, int ) {
 //    qDebug( "WavPlugin::audioReadSamples" );
     return d->add( output, samples, samplesMade, channels != 1 );
 }
 
 double WavPlugin::getTime() {
-//    qDebug( "WavPlugin::getTime" );
-    return 0.0;
+//    qDebug( "WavPlugin::getTime" ); //this is a stupid hack here!!
+  return d->chunkdata.wBitsPerSample; /*0.0*/;
 }
 
-int WavPlugin::audioBitsPerSample( int ) {
-//    qDebug( "WavPlugin::audioFormat %d", d->chunkdata.wBitsPerSample );
-    return d->chunkdata.wBitsPerSample; // 
-}
+// int WavPlugin::audioBitsPerSample( int ) {
+// //    qDebug( "WavPlugin::audioFormat %d", d->chunkdata.wBitsPerSample );
+//     return d->chunkdata.wBitsPerSample; // 
+// }