renamed rate to samplerate.

3 files changed, 10 insertions, 10 deletions

diff --git a/avi2swf/v2swf.c b/avi2swf/v2swf.c
index e983c36a..2caed7b8 100644
--- a/avi2swf/v2swf.c
+++ b/avi2swf/v2swf.c
@@ -168,7 +168,7 @@ static void writeShape(v2swf_internal_t*i, int id, int gfxid, int width, int hei
 static int getSamples(videoreader_t*video, S16*data, int len, double speedup)
 {
     double pos = 0;
-    double ratio = (double) video->rate * speedup / swf_mp3_in_samplerate;
+    double ratio = (double) video->samplerate * speedup / swf_mp3_in_samplerate;
     int rlen = (int)(len * ratio);
     int t;
     S16 tmp[576*32];
@@ -209,7 +209,7 @@ static void writeAudioForOneFrame(v2swf_internal_t* i)
 
     msg("writeAudioForOneFrame()");
 
-    if(i->video->channels<=0 || i->video->rate<=0)
+    if(i->video->channels<=0 || i->video->samplerate<=0)
 	return; /* no sound in video */
 
     blocksize = (i->samplerate > 22050) ? 1152 : 576;
@@ -229,7 +229,7 @@ static void writeAudioForOneFrame(v2swf_internal_t* i)
 	/* The pre-processing of sound samples in getSamples(..) above
 	   re-samples the sound to swf_mp3_in_samplerate. It is best to
 	   simply make it the original samplerate:  */
-	swf_mp3_in_samplerate = i->video->rate;
+	swf_mp3_in_samplerate = i->video->samplerate;
 
 	/* first run - initialize */
 	swf_mp3_channels = 1;//i->video->channels;
@@ -268,7 +268,7 @@ static void writeAudioForOneFrame(v2swf_internal_t* i)
     /* write num frames, max 1 block */
     for(pos=0;pos<num;pos++) {
         if(!getSamples(i->video, block1, blocksize * (double)swf_mp3_in_samplerate/swf_mp3_out_samplerate, speedup)) {
-	    i->video->rate = i->video->channels = 0; //end of soundtrack
+	    i->video->samplerate = i->video->channels = 0; //end of soundtrack
 	    return;
 	}
 	if(!pos) {
diff --git a/avi2swf/videoreader.h b/avi2swf/videoreader.h
index 73799a2a..1924eafc 100644
--- a/avi2swf/videoreader.h
+++ b/avi2swf/videoreader.h
@@ -18,7 +18,7 @@ typedef struct _videoreader_t
 
     /* audio */
     int channels;
-    int rate;
+    int samplerate;
 
     /* progress */
     int frame;
diff --git a/avi2swf/videoreader_avifile.cc b/avi2swf/videoreader_avifile.cc
index 653412f0..0cd2599b 100644
--- a/avi2swf/videoreader_avifile.cc
+++ b/avi2swf/videoreader_avifile.cc
@@ -269,7 +269,7 @@ int videoreader_avifile_open(videoreader_t* v, char* filename)
     v->height = head.dwHeight;
     dwMicroSecPerFrame = head.dwMicroSecPerFrame;
     samplesperframe = astream->GetEndPos()/astream->GetEndTime()*head.dwMicroSecPerFrame/1000000;
-    v->rate = (int)(astream->GetEndPos()/astream->GetEndTime());
+    v->samplerate = (int)(astream->GetEndPos()/astream->GetEndTime());
     v->fps = 1000000.0/dwMicroSecPerFrame;
     i->soundbits = 16;
 #else
@@ -290,11 +290,11 @@ int videoreader_avifile_open(videoreader_t* v, char* filename)
 	audioinfo = i->astream->GetStreamInfo();
 
 	v->channels = wave.nChannels;
-	v->rate = wave.nSamplesPerSec;
+	v->samplerate = wave.nSamplesPerSec;
 	i->soundbits = wave.wBitsPerSample;
 
-	if(v->channels==0 || v->rate==0 || i->soundbits==0 || wave.wFormatTag!=1) {
-	    v->rate = audioinfo->GetAudioSamplesPerSec();
+	if(v->channels==0 || v->samplerate==0 || i->soundbits==0 || wave.wFormatTag!=1) {
+	    v->samplerate = audioinfo->GetAudioSamplesPerSec();
 	    v->channels = audioinfo->GetAudioChannels();
 	    i->soundbits = audioinfo->GetAudioBitsPerSample();
 	}
@@ -308,7 +308,7 @@ int videoreader_avifile_open(videoreader_t* v, char* filename)
 	    i->do_audio = 0;
 	    i->soundbits = 0;
 	    v->channels = 0;
-	    v->rate = 0;
+	    v->samplerate = 0;
 	}
     }
 #endif