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| author | Warren Dukes <warren.dukes@gmail.com> | 2004-03-22 02:44:22 +0000 |
|---|---|---|
| committer | Warren Dukes <warren.dukes@gmail.com> | 2004-03-22 02:44:22 +0000 |
| commit | 145f0b62d26974c5e15ad65c902045fe671a9d05 (patch) | |
| tree | f86617eeae1a30f1e97a35507b93756a99cf0b29 /src/mp4_decode.c | |
| parent | 4c1eb9225d5a741e1234d48eb38a8df3da908259 (diff) | |
| download | mpd-145f0b62d26974c5e15ad65c902045fe671a9d05.tar.gz mpd-145f0b62d26974c5e15ad65c902045fe671a9d05.tar.xz mpd-145f0b62d26974c5e15ad65c902045fe671a9d05.zip | |
aac decoding!
need to work on seeking for AAC
also, don't reset cb->begin on seek to 0,
instead just set cb->end=cb->begin, works much better for
disabling seeking (like ADIF AAC's)
git-svn-id: https://svn.musicpd.org/mpd/trunk@356 09075e82-0dd4-0310-85a5-a0d7c8717e4f
Diffstat (limited to '')
| -rw-r--r-- | src/mp4_decode.c | 8 |
1 files changed, 3 insertions, 5 deletions
diff --git a/src/mp4_decode.c b/src/mp4_decode.c index a2cfc2ee0..d3c16b73e 100644 --- a/src/mp4_decode.c +++ b/src/mp4_decode.c @@ -2,8 +2,6 @@ * (c)2003-2004 by Warren Dukes (shank@mercury.chem.pitt.edu) * This project's homepage is: http://www.musicpd.org * - * libaudiofile (wave) support added by Eric Wong <normalperson@yhbt.net> - * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License as published by * the Free Software Foundation; either version 2 of the License, or @@ -149,7 +147,7 @@ int mp4_decode(Buffer * cb, AudioFormat * af, DecoderControl * dc) { if(faacDecInit2(decoder,mp4Buffer,mp4BufferSize,&sampleRate,&channels) < 0) { - ERROR("Error initializing AAC decoder library.\n"); + ERROR("Error not a AAC stream.\n"); faacDecClose(decoder); mp4ff_close(mp4fh); free(mp4cb); @@ -210,7 +208,7 @@ int mp4_decode(Buffer * cb, AudioFormat * af, DecoderControl * dc) { if(dc->seek && seekPositionFound) { seekPositionFound = 0; chunkLen = 0; - cb->end = 0; + cb->end = cb->begin; cb->wrap = 0; dc->seek = 0; } @@ -233,7 +231,7 @@ int mp4_decode(Buffer * cb, AudioFormat * af, DecoderControl * dc) { } if(channels*(dur+offset) > frameInfo.samples) { - dur = frameInfo.samples; + dur = frameInfo.samples/channels; offset = 0; } |