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authorAlex Converse <alex.converse@gmail.com>2009-07-08 20:01:31 +0000
committerAlex Converse <alex.converse@gmail.com>2009-07-08 20:01:31 +0000
commit78e65cd7726942a1615ead039abe0bfa79341212 (patch)
tree7003e32f0234d3fb6d7959e9f193e2ec733df5c6 /libavcodec/aacenc.h
parent5e039e1b4c0fe25c76faa7ea107db60264edb757 (diff)
Merge the AAC encoder from SoC svn. It is still considered experimental.
Originally committed as revision 19375 to svn://svn.ffmpeg.org/ffmpeg/trunk
Diffstat (limited to 'libavcodec/aacenc.h')
-rw-r--r--libavcodec/aacenc.h71
1 files changed, 71 insertions, 0 deletions
diff --git a/libavcodec/aacenc.h b/libavcodec/aacenc.h
new file mode 100644
index 0000000000..458d3d5d99
--- /dev/null
+++ b/libavcodec/aacenc.h
@@ -0,0 +1,71 @@
+/*
+ * AAC encoder
+ * Copyright (C) 2008 Konstantin Shishkov
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#ifndef AVCODEC_AACENC_H
+#define AVCODEC_AACENC_H
+
+#include "avcodec.h"
+#include "put_bits.h"
+#include "dsputil.h"
+
+#include "aac.h"
+
+#include "psymodel.h"
+
+struct AACEncContext;
+
+typedef struct AACCoefficientsEncoder{
+ void (*search_for_quantizers)(AVCodecContext *avctx, struct AACEncContext *s,
+ SingleChannelElement *sce, const float lambda);
+ void (*encode_window_bands_info)(struct AACEncContext *s, SingleChannelElement *sce,
+ int win, int group_len, const float lambda);
+ void (*quantize_and_encode_band)(struct AACEncContext *s, PutBitContext *pb, const float *in, int size,
+ int scale_idx, int cb, const float lambda);
+ void (*search_for_ms)(struct AACEncContext *s, ChannelElement *cpe, const float lambda);
+}AACCoefficientsEncoder;
+
+extern AACCoefficientsEncoder ff_aac_coders[];
+
+/**
+ * AAC encoder context
+ */
+typedef struct AACEncContext {
+ PutBitContext pb;
+ MDCTContext mdct1024; ///< long (1024 samples) frame transform context
+ MDCTContext mdct128; ///< short (128 samples) frame transform context
+ DSPContext dsp;
+ DECLARE_ALIGNED_16(FFTSample, output[2048]); ///< temporary buffer for MDCT input coefficients
+ int16_t* samples; ///< saved preprocessed input
+
+ int samplerate_index; ///< MPEG-4 samplerate index
+
+ ChannelElement *cpe; ///< channel elements
+ FFPsyContext psy;
+ struct FFPsyPreprocessContext* psypp;
+ AACCoefficientsEncoder *coder;
+ int cur_channel;
+ int last_frame;
+ float lambda;
+ DECLARE_ALIGNED_16(int, qcoefs[96][2]); ///< quantized coefficients
+ DECLARE_ALIGNED_16(float, scoefs[1024]); ///< scaled coefficients
+} AACEncContext;
+
+#endif /* AVCODEC_AACENC_H */