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libavcodec
psymodel.c
Go to the documentation of this file.
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/*
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* audio encoder psychoacoustic model
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* Copyright (C) 2008 Konstantin Shishkov
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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#include <string.h>
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#include "
avcodec.h
"
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#include "
psymodel.h
"
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#include "
iirfilter.h
"
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#include "
libavutil/mem.h
"
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extern
const
FFPsyModel
ff_aac_psy_model
;
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av_cold
int
ff_psy_init
(
FFPsyContext
*ctx,
AVCodecContext
*avctx,
int
num_lens,
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const
uint8_t
**bands,
const
int
* num_bands,
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int
num_groups,
const
uint8_t
*group_map)
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{
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int
i, j, k = 0;
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ctx->
avctx
= avctx;
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ctx->
ch
=
av_mallocz
(
sizeof
(ctx->
ch
[0]) * avctx->
channels
* 2);
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ctx->
group
=
av_mallocz
(
sizeof
(ctx->
group
[0]) * num_groups);
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ctx->
bands
=
av_malloc
(
sizeof
(ctx->
bands
[0]) * num_lens);
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ctx->
num_bands
=
av_malloc
(
sizeof
(ctx->
num_bands
[0]) * num_lens);
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memcpy(ctx->
bands
, bands,
sizeof
(ctx->
bands
[0]) * num_lens);
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memcpy(ctx->
num_bands
, num_bands,
sizeof
(ctx->
num_bands
[0]) * num_lens);
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/* assign channels to groups (with virtual channels for coupling) */
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for
(i = 0; i < num_groups; i++) {
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/* NOTE: Add 1 to handle the AAC chan_config without modification.
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* This has the side effect of allowing an array of 0s to map
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* to one channel per group.
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*/
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ctx->
group
[i].
num_ch
= group_map[i] + 1;
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for
(j = 0; j < ctx->
group
[i].
num_ch
* 2; j++)
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ctx->
group
[i].
ch
[j] = &ctx->
ch
[k++];
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}
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switch
(ctx->
avctx
->
codec_id
) {
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case
AV_CODEC_ID_AAC
:
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ctx->
model
= &
ff_aac_psy_model
;
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break
;
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}
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if
(ctx->
model
->
init
)
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return
ctx->
model
->
init
(ctx);
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return
0;
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}
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FFPsyChannelGroup
*
ff_psy_find_group
(
FFPsyContext
*ctx,
int
channel)
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{
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int
i = 0, ch = 0;
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while
(ch <= channel)
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ch += ctx->
group
[i++].
num_ch
;
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return
&ctx->
group
[i-1];
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}
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av_cold
void
ff_psy_end
(
FFPsyContext
*ctx)
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{
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if
(ctx->
model
->
end
)
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ctx->
model
->
end
(ctx);
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av_freep
(&ctx->
bands
);
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av_freep
(&ctx->
num_bands
);
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av_freep
(&ctx->
group
);
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av_freep
(&ctx->
ch
);
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}
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typedef
struct
FFPsyPreprocessContext
{
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AVCodecContext
*
avctx
;
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float
stereo_att
;
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struct
FFIIRFilterCoeffs
*
fcoeffs
;
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struct
FFIIRFilterState
**
fstate
;
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}
FFPsyPreprocessContext
;
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#define FILT_ORDER 4
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av_cold
struct
FFPsyPreprocessContext
*
ff_psy_preprocess_init
(
AVCodecContext
*
avctx
)
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{
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FFPsyPreprocessContext
*ctx;
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int
i;
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float
cutoff_coeff = 0;
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ctx =
av_mallocz
(
sizeof
(
FFPsyPreprocessContext
));
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ctx->
avctx
=
avctx
;
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if
(avctx->
cutoff
> 0)
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cutoff_coeff = 2.0 * avctx->
cutoff
/ avctx->
sample_rate
;
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if
(!cutoff_coeff && avctx->
codec_id
==
AV_CODEC_ID_AAC
)
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cutoff_coeff = 2.0 *
AAC_CUTOFF
(avctx) / avctx->
sample_rate
;
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if
(cutoff_coeff && cutoff_coeff < 0.98)
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ctx->
fcoeffs
=
ff_iir_filter_init_coeffs
(avctx,
FF_FILTER_TYPE_BUTTERWORTH
,
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FF_FILTER_MODE_LOWPASS
,
FILT_ORDER
,
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cutoff_coeff, 0.0, 0.0);
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if
(ctx->
fcoeffs
) {
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ctx->
fstate
=
av_mallocz
(
sizeof
(ctx->
fstate
[0]) * avctx->
channels
);
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for
(i = 0; i < avctx->
channels
; i++)
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ctx->
fstate
[i] =
ff_iir_filter_init_state
(
FILT_ORDER
);
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}
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return
ctx;
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}
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void
ff_psy_preprocess
(
struct
FFPsyPreprocessContext
*ctx,
float
**audio,
int
channels)
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{
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int
ch;
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int
frame_size
= ctx->
avctx
->
frame_size
;
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if
(ctx->
fstate
) {
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for
(ch = 0; ch < channels; ch++)
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ff_iir_filter_flt
(ctx->
fcoeffs
, ctx->
fstate
[ch], frame_size,
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&audio[ch][frame_size], 1, &audio[ch][frame_size], 1);
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}
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}
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av_cold
void
ff_psy_preprocess_end
(
struct
FFPsyPreprocessContext
*ctx)
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{
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int
i;
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ff_iir_filter_free_coeffs
(ctx->
fcoeffs
);
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if
(ctx->
fstate
)
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for
(i = 0; i < ctx->
avctx
->
channels
; i++)
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ff_iir_filter_free_state
(ctx->
fstate
[i]);
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av_freep
(&ctx->
fstate
);
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av_free
(ctx);
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}
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