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when converting from bps -> kbps, divide by 1000, not 1024 

git-svn-id: https://svn.musicpd.org/mpd/trunk@592 09075e82-0dd4-0310-85a5-a0d7c8717e4f
This commit is contained in:
Warren Dukes 2004-04-03 04:01:31 +00:00
parent 785cdb0114
commit 73bcce335f
6 changed files with 6 additions and 6 deletions
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2
src/aac_decode.c
View File

@ -357,7 +357,7 @@ int aac_decode(Buffer * cb, AudioFormat * af, DecoderControl * dc) {
if(sampleCount>0) {
bitRate = frameInfo.bytesconsumed*8.0*
frameInfo.channels*sampleRate/
frameInfo.samples/1024+0.5;
frameInfo.samples/1000+0.5;
time+= (float)(frameInfo.samples)/frameInfo.channels/
sampleRate;
}

2
src/audiofile_decode.c
View File

@ -79,7 +79,7 @@ int audiofile_decode(Buffer * cb, AudioFormat * af, DecoderControl * dc)
cb->totalTime = ((float)frame_count/(float)af->sampleRate);
bitRate = st.st_size*8.0/cb->totalTime/1024.0+0.5;
bitRate = st.st_size*8.0/cb->totalTime/1000.0+0.5;
if (af->bits != 8 && af->bits != 16) {
ERROR("Only 8 and 16-bit files are supported. %s is %i-bit\n",

2
src/flac_decode.c
View File

@ -219,7 +219,7 @@ FLAC__StreamDecoderWriteStatus flacWrite(const FLAC__FileDecoder *dec, const FLA
FLAC__file_decoder_get_decode_position(dec,&newPosition);
if(data->position) {
data->bitRate = ((newPosition-data->position)*8.0/timeChange)
/1024+0.5;
/1000+0.5;
}
data->position = newPosition;

2
src/mp3_decode.c
View File

@ -416,7 +416,7 @@ int mp3ChildSendData(mp3DecodeData * data, Buffer * cb, DecoderControl * dc) {
memcpy(cb->chunks+cb->end*CHUNK_SIZE,data->outputBuffer,CHUNK_SIZE);
cb->chunkSize[cb->end] = data->outputPtr-data->outputBuffer;
cb->bitRate[cb->end] = data->bitRate/1024;
cb->bitRate[cb->end] = data->bitRate/1000;
cb->times[cb->end] = data->elapsedTime;
cb->end++;

2
src/mp4_decode.c
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@ -270,7 +270,7 @@ int mp4_decode(Buffer * cb, AudioFormat * af, DecoderControl * dc) {
initial =0;
bitRate = frameInfo.bytesconsumed*8.0*
frameInfo.channels*scale/
frameInfo.samples/1024+0.5;
frameInfo.samples/1000+0.5;
}

2
src/ogg_decode.c
View File

@ -122,7 +122,7 @@ int ogg_decode(Buffer * cb, AudioFormat * af, DecoderControl * dc)
chunkpos = 0;
cb->times[cb->end] = ov_time_tell(&vf);
if((test = ov_bitrate_instant(&vf))>0) {
bitRate = test/1024;
bitRate = test/1000;
}
cb->bitRate[cb->end] = bitRate;
cb->end++;