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author | Justin Ruggles <justin.ruggles@gmail.com> | 2011-01-30 15:06:46 +0000 |
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committer | Michael Niedermayer <michaelni@gmx.at> | 2011-02-04 03:08:09 +0100 |
commit | fe2ff6d24745f0739bfde9061092c1268557310b (patch) | |
tree | 9cbcf8b2472dd7612dd84c8b6b237d9d02b4daf9 /libavcodec/aacdec.c | |
parent | a35d782d28ef0497f2b65eb300c2e6a6028fc165 (diff) | |
download | ffmpeg-fe2ff6d24745f0739bfde9061092c1268557310b.tar.gz |
Separate format conversion DSP functions from DSPContext.
This will be beneficial for use with the audio conversion API without
requiring it to depend on all of dsputil.
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit c73d99e672329c8f2df290736ffc474c360ac4ae)
Diffstat (limited to 'libavcodec/aacdec.c')
-rw-r--r-- | libavcodec/aacdec.c | 4 |
1 files changed, 3 insertions, 1 deletions
diff --git a/libavcodec/aacdec.c b/libavcodec/aacdec.c index 0ea7dc84a5..411c1dfc1b 100644 --- a/libavcodec/aacdec.c +++ b/libavcodec/aacdec.c @@ -85,6 +85,7 @@ #include "get_bits.h" #include "dsputil.h" #include "fft.h" +#include "fmtconvert.h" #include "lpc.h" #include "aac.h" @@ -562,6 +563,7 @@ static av_cold int aac_decode_init(AVCodecContext *avctx) ff_aac_sbr_init(); dsputil_init(&ac->dsp, avctx); + ff_fmt_convert_init(&ac->fmt_conv, avctx); ac->random_state = 0x1f2e3d4c; @@ -2032,7 +2034,7 @@ static int aac_decode_frame_int(AVCodecContext *avctx, void *data, *data_size = data_size_tmp; if (samples) - ac->dsp.float_to_int16_interleave(data, (const float **)ac->output_data, samples, avctx->channels); + ac->fmt_conv.float_to_int16_interleave(data, (const float **)ac->output_data, samples, avctx->channels); if (ac->output_configured) ac->output_configured = OC_LOCKED; |