From 2ec1c5ff3c1ff67825fb449c9eab2c3e4ff441f6 Mon Sep 17 00:00:00 2001 From: Warren Dukes Date: Mon, 10 May 2004 12:35:18 +0000 Subject: some more work on organizing code for resampling/audioFormat conversion git-svn-id: https://svn.musicpd.org/mpd/trunk@968 09075e82-0dd4-0310-85a5-a0d7c8717e4f --- src/aac_decode.c | 12 +++++++----- 1 file changed, 7 insertions(+), 5 deletions(-) (limited to 'src/aac_decode.c') diff --git a/src/aac_decode.c b/src/aac_decode.c index 24171adb7..7013502d4 100644 --- a/src/aac_decode.c +++ b/src/aac_decode.c @@ -251,7 +251,7 @@ int getAacTotalTime(char * file) { } -int aac_decode(OutputBuffer * cb, AudioFormat * af, DecoderControl * dc) { +int aac_decode(OutputBuffer * cb, DecoderControl * dc) { float time; float totalTime; faacDecHandle decoder; @@ -306,9 +306,9 @@ int aac_decode(OutputBuffer * cb, AudioFormat * af, DecoderControl * dc) { return -1; } - af->bits = 16; + dc->audioFormat.bits = 16; - cb->totalTime = totalTime; + dc->totalTime = totalTime; time = 0.0; @@ -342,8 +342,10 @@ int aac_decode(OutputBuffer * cb, AudioFormat * af, DecoderControl * dc) { #endif if(dc->start) { - af->channels = frameInfo.channels; - af->sampleRate = sampleRate; + dc->audioFormat.channels = frameInfo.channels; + dc->audioFormat.sampleRate = sampleRate; + getOutputAudioFormat(&(dc->audioFormat), + &(cb->audioFormat)); dc->state = DECODE_STATE_DECODE; dc->start = 0; } -- cgit v1.2.3