From: kramm Date: Mon, 20 Sep 2004 16:11:16 +0000 (+0000) Subject: renamed rate to samplerate. X-Git-Tag: release-0-6-2~61 X-Git-Url: http://git.asbjorn.biz/?p=swftools.git;a=commitdiff_plain;h=d6b5ebb01904d445f6ed9e00a54ca5e1e6c800cd renamed rate to samplerate. --- diff --git a/avi2swf/v2swf.c b/avi2swf/v2swf.c index e983c36..2caed7b 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;posvideo, 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 73799a2..1924eaf 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 653412f..0cd2599 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