From 73bcce335f49df5b919ba326765a94fd821561ea Mon Sep 17 00:00:00 2001 From: Warren Dukes Date: Sat, 3 Apr 2004 04:01:31 +0000 Subject: 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 --- src/aac_decode.c | 2 +- src/audiofile_decode.c | 2 +- src/flac_decode.c | 2 +- src/mp3_decode.c | 2 +- src/mp4_decode.c | 2 +- src/ogg_decode.c | 2 +- 6 files changed, 6 insertions(+), 6 deletions(-) diff --git a/src/aac_decode.c b/src/aac_decode.c index 0610b774a..1c0f867d9 100644 --- a/src/aac_decode.c +++ b/src/aac_decode.c @@ -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; } diff --git a/src/audiofile_decode.c b/src/audiofile_decode.c index d34c029ee..7cd193581 100644 --- a/src/audiofile_decode.c +++ b/src/audiofile_decode.c @@ -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", diff --git a/src/flac_decode.c b/src/flac_decode.c index afb5bcf74..25857cb6e 100644 --- a/src/flac_decode.c +++ b/src/flac_decode.c @@ -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; diff --git a/src/mp3_decode.c b/src/mp3_decode.c index 618f6f5ef..84192078b 100644 --- a/src/mp3_decode.c +++ b/src/mp3_decode.c @@ -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++; diff --git a/src/mp4_decode.c b/src/mp4_decode.c index 73e99215a..77e98b676 100644 --- a/src/mp4_decode.c +++ b/src/mp4_decode.c @@ -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; } diff --git a/src/ogg_decode.c b/src/ogg_decode.c index e5a9d58d5..ef74637d9 100644 --- a/src/ogg_decode.c +++ b/src/ogg_decode.c @@ -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++; -- cgit v1.2.3