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mirror of https://github.com/FFmpeg/FFmpeg.git synced 2024-12-07 11:13:41 +02:00
FFmpeg/libavcodec/psymodel.c
Michael Niedermayer c581cb4e4f Merge remote-tracking branch 'qatar/master'
* qatar/master:
  Fix even more missing includes after the common.h removal
  build: Factor out rangecoder dependencies to CONFIG_RANGECODER
  build: Factor out error resilience dependencies to CONFIG_ERROR_RESILIENCE
  x86: avcodec: Consistently name all init files
  Add more missing includes after removing the implicit common.h
  Add some more missing includes after removing the implicit common.h
  Don't include common.h from avutil.h
  rtmp: Automatically compute the hash for SWFVerification

Conflicts:
	configure
	doc/APIchanges
	doc/examples/decoding_encoding.c
	libavcodec/Makefile
	libavcodec/assdec.c
	libavcodec/audio_frame_queue.c
	libavcodec/avpacket.c
	libavcodec/dv_profile.c
	libavcodec/dwt.c
	libavcodec/libtheoraenc.c
	libavcodec/rawdec.c
	libavcodec/rv40dsp.c
	libavcodec/tiff.c
	libavcodec/tiffenc.c
	libavcodec/v210dec.h
	libavcodec/vc1dsp.c
	libavcodec/x86/Makefile
	libavfilter/asrc_anullsrc.c
	libavfilter/avfilter.c
	libavfilter/buffer.c
	libavfilter/formats.c
	libavfilter/vf_ass.c
	libavfilter/vf_drawtext.c
	libavfilter/vf_fade.c
	libavfilter/vf_select.c
	libavfilter/video.c
	libavfilter/vsrc_testsrc.c
	libavformat/version.h
	libavutil/audioconvert.c
	libavutil/error.h
	libavutil/version.h

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2012-08-16 16:20:30 +02:00

143 lines
4.5 KiB
C

/*
* audio encoder psychoacoustic model
* 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
*/
#include <string.h>
#include "avcodec.h"
#include "psymodel.h"
#include "iirfilter.h"
#include "libavutil/mem.h"
extern const FFPsyModel ff_aac_psy_model;
av_cold int ff_psy_init(FFPsyContext *ctx, AVCodecContext *avctx, int num_lens,
const uint8_t **bands, const int* num_bands,
int num_groups, const uint8_t *group_map)
{
int i, j, k = 0;
ctx->avctx = avctx;
ctx->ch = av_mallocz(sizeof(ctx->ch[0]) * avctx->channels * 2);
ctx->group = av_mallocz(sizeof(ctx->group[0]) * num_groups);
ctx->bands = av_malloc (sizeof(ctx->bands[0]) * num_lens);
ctx->num_bands = av_malloc (sizeof(ctx->num_bands[0]) * num_lens);
memcpy(ctx->bands, bands, sizeof(ctx->bands[0]) * num_lens);
memcpy(ctx->num_bands, num_bands, sizeof(ctx->num_bands[0]) * num_lens);
/* assign channels to groups (with virtual channels for coupling) */
for (i = 0; i < num_groups; i++) {
/* NOTE: Add 1 to handle the AAC chan_config without modification.
* This has the side effect of allowing an array of 0s to map
* to one channel per group.
*/
ctx->group[i].num_ch = group_map[i] + 1;
for (j = 0; j < ctx->group[i].num_ch * 2; j++)
ctx->group[i].ch[j] = &ctx->ch[k++];
}
switch (ctx->avctx->codec_id) {
case AV_CODEC_ID_AAC:
ctx->model = &ff_aac_psy_model;
break;
}
if (ctx->model->init)
return ctx->model->init(ctx);
return 0;
}
FFPsyChannelGroup *ff_psy_find_group(FFPsyContext *ctx, int channel)
{
int i = 0, ch = 0;
while (ch <= channel)
ch += ctx->group[i++].num_ch;
return &ctx->group[i-1];
}
av_cold void ff_psy_end(FFPsyContext *ctx)
{
if (ctx->model->end)
ctx->model->end(ctx);
av_freep(&ctx->bands);
av_freep(&ctx->num_bands);
av_freep(&ctx->group);
av_freep(&ctx->ch);
}
typedef struct FFPsyPreprocessContext{
AVCodecContext *avctx;
float stereo_att;
struct FFIIRFilterCoeffs *fcoeffs;
struct FFIIRFilterState **fstate;
}FFPsyPreprocessContext;
#define FILT_ORDER 4
av_cold struct FFPsyPreprocessContext* ff_psy_preprocess_init(AVCodecContext *avctx)
{
FFPsyPreprocessContext *ctx;
int i;
float cutoff_coeff = 0;
ctx = av_mallocz(sizeof(FFPsyPreprocessContext));
ctx->avctx = avctx;
if (avctx->cutoff > 0)
cutoff_coeff = 2.0 * avctx->cutoff / avctx->sample_rate;
if (!cutoff_coeff && avctx->codec_id == AV_CODEC_ID_AAC)
cutoff_coeff = 2.0 * AAC_CUTOFF(avctx) / avctx->sample_rate;
if (cutoff_coeff)
ctx->fcoeffs = ff_iir_filter_init_coeffs(avctx, FF_FILTER_TYPE_BUTTERWORTH,
FF_FILTER_MODE_LOWPASS, FILT_ORDER,
cutoff_coeff, 0.0, 0.0);
if (ctx->fcoeffs) {
ctx->fstate = av_mallocz(sizeof(ctx->fstate[0]) * avctx->channels);
for (i = 0; i < avctx->channels; i++)
ctx->fstate[i] = ff_iir_filter_init_state(FILT_ORDER);
}
return ctx;
}
void ff_psy_preprocess(struct FFPsyPreprocessContext *ctx, float **audio, int channels)
{
int ch;
int frame_size = ctx->avctx->frame_size;
if (ctx->fstate) {
for (ch = 0; ch < channels; ch++)
ff_iir_filter_flt(ctx->fcoeffs, ctx->fstate[ch], frame_size,
&audio[ch][frame_size], 1, &audio[ch][frame_size], 1);
}
}
av_cold void ff_psy_preprocess_end(struct FFPsyPreprocessContext *ctx)
{
int i;
ff_iir_filter_free_coeffs(ctx->fcoeffs);
if (ctx->fstate)
for (i = 0; i < ctx->avctx->channels; i++)
ff_iir_filter_free_state(ctx->fstate[i]);
av_freep(&ctx->fstate);
av_free(ctx);
}