update wavfile

1 files changed, 20 insertions, 15 deletions

diff --git a/noncore/multimedia/opierec/wavFile.cpp b/noncore/multimedia/opierec/wavFile.cpp
index 7bfffb6..b177c91 100644
--- a/noncore/multimedia/opierec/wavFile.cpp
+++ b/noncore/multimedia/opierec/wavFile.cpp
@@ -24,9 +24,9 @@
 WavFile::WavFile( QObject * parent,const QString &fileName, bool makeNwFile, int sampleRate,
                   int channels, int resolution, int format )
         : QObject( parent)
 {
-qDebug("new wave file");
+//qDebug("new wave file");
     bool b =  makeNwFile;
     wavSampleRate=sampleRate;
     wavFormat=format;
     wavChannels=channels;
@@ -40,9 +40,9 @@ qDebug("new wave file");
 }
 
 bool WavFile::newFile() {
 
-  qDebug("Set up new file");
+//  qDebug("Set up new file");
   Config cfg("OpieRec");
   cfg.setGroup("Settings");
 
   currentFileName=cfg.readEntry("directory",QDir::homeDirPath());
@@ -59,19 +59,24 @@ bool WavFile::newFile() {
   else
     currentFileName += date;
   currentFileName+=".wav";
 
-  qDebug("set up file for recording: "+currentFileName);
-  char *pointer;
+//  qDebug("set up file for recording: "+currentFileName);
+  char pointer[] = "/tmp/opierec-XXXXXX";
+  int fd = 0;
 
   if( currentFileName.find("/mnt",0,true) == -1
       && currentFileName.find("/tmp",0,true) == -1 ) {
     // if destination file is most likely in flash (assuming jffs2)
     // we have to write to a different filesystem first
         
     useTmpFile = true;
-    pointer=tmpnam(NULL); 
-    qDebug("Opening tmp file %s",pointer);
+    if(( fd = mkstemp( pointer)) < 0 ) {
+       perror("mkstemp failed");
+       return false;
+    }
+    
+//    qDebug("Opening tmp file %s",pointer);
     track.setName( pointer);
 
   } else { //just use regular file.. no moving
       
@@ -100,9 +105,9 @@ void WavFile::closeFile() {
         track.close();
 }
 
 int WavFile::openFile(const QString &currentFileName) {
-    qDebug("open play file "+currentFileName);
+//    qDebug("open play file "+currentFileName);
     closeFile();
     
   track.setName(currentFileName);
 
@@ -125,13 +130,13 @@ bool WavFile::setWavHeader(int fd, wavhdr *hdr) {
   (*hdr).fmtLen = 16; // format length = 16
 
   if( wavFormat == WAVE_FORMAT_PCM) {
     (*hdr).fmtTag = 1; // PCM
-    qDebug("set header  WAVE_FORMAT_PCM");
+//    qDebug("set header  WAVE_FORMAT_PCM");
   }
   else {
     (*hdr).fmtTag = WAVE_FORMAT_DVI_ADPCM; //intel ADPCM
-    qDebug("set header  WAVE_FORMAT_DVI_ADPCM");
+ //    qDebug("set header  WAVE_FORMAT_DVI_ADPCM");
   }
 
   //  (*hdr).nChannels = 1;//filePara.channels;// ? 2 : 1*/; // channels
   (*hdr).nChannels = wavChannels;// ? 2 : 1*/; // channels
@@ -143,10 +148,10 @@ bool WavFile::setWavHeader(int fd, wavhdr *hdr) {
 
   strncpy((*hdr).dataID, "data", 4);
 
   write( fd,hdr, sizeof(*hdr));
-  qDebug("writing header: bitrate%d, samplerate %d,  channels %d",
-         wavResolution, wavSampleRate, wavChannels);
+//   qDebug("writing header: bitrate%d, samplerate %d,  channels %d",
+//          wavResolution, wavSampleRate, wavChannels);
   return true;
 }
 
 bool WavFile::adjustHeaders(int fd, int total) {
@@ -154,14 +159,14 @@ bool WavFile::adjustHeaders(int fd, int total) {
   int i = total + 36;
   write( fd, &i, sizeof(i));
   lseek( fd, 40, SEEK_SET);
   write( fd, &total, sizeof(total));
-  qDebug("adjusting header %d", total);
+//  qDebug("adjusting header %d", total);
   return true;
 }
 
 int WavFile::parseWavHeader(int fd) {
-  qDebug("Parsing wav header");
+//  qDebug("Parsing wav header");
   char string[4];
   int found;
   short fmt;
   unsigned short ch, bitrate;
@@ -257,10 +262,10 @@ int WavFile::parseWavHeader(int fd) {
         qDebug("Could not read from sound file.\n");
         return -1;
       } else {
         wavNumberSamples =  longdata;
-         qDebug("file has length of %d \nlasting %d seconds", longdata,
-        (( longdata / wavSampleRate) / wavChannels) / ( wavChannels*( wavResolution/8)) );
+         qDebug("file has length of %d \nlasting %d seconds", (int)longdata,
+        (int)(( longdata / wavSampleRate) / wavChannels) / ( wavChannels*( wavResolution/8)) );
 //        wavSeconds = (( longdata / wavSampleRate) / wavChannels) / ( wavChannels*( wavResolution/8));
         
   return longdata;
       }