Rename ffmpeg-4.2-fit to ffmpeg-4-fit

trunk/3rdparty/ffmpeg-4-fit/libavcodec/psymodel.c

@@ -0,0 +1,161 @@
+/*
+ * 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_array(sizeof(ctx->ch[0]), avctx->channels * 2);
+    ctx->group     = av_mallocz_array(sizeof(ctx->group[0]), num_groups);
+    ctx->bands     = av_malloc_array (sizeof(ctx->bands[0]),      num_lens);
+    ctx->num_bands = av_malloc_array (sizeof(ctx->num_bands[0]),  num_lens);
+    ctx->cutoff    = avctx->cutoff;
+
+    if (!ctx->ch || !ctx->group || !ctx->bands || !ctx->num_bands) {
+        ff_psy_end(ctx);
+        return AVERROR(ENOMEM);
+    }
+
+    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 && 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;
+    struct FFIIRFilterContext fiir;
+}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));
+    if (!ctx)
+        return NULL;
+    ctx->avctx = avctx;
+
+    /* AAC has its own LP method */
+    if (avctx->codec_id != AV_CODEC_ID_AAC) {
+        if (avctx->cutoff > 0)
+            cutoff_coeff = 2.0 * avctx->cutoff / avctx->sample_rate;
+
+        if (cutoff_coeff && cutoff_coeff < 0.98)
+        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_array(sizeof(ctx->fstate[0]), avctx->channels);
+            if (!ctx->fstate) {
+                av_free(ctx->fcoeffs);
+                av_free(ctx);
+                return NULL;
+            }
+            for (i = 0; i < avctx->channels; i++)
+                ctx->fstate[i] = ff_iir_filter_init_state(FILT_ORDER);
+        }
+    }
+
+    ff_iir_filter_init(&ctx->fiir);
+
+    return ctx;
+}
+
+void ff_psy_preprocess(struct FFPsyPreprocessContext *ctx, float **audio, int channels)
+{
+    int ch;
+    int frame_size = ctx->avctx->frame_size;
+    FFIIRFilterContext *iir = &ctx->fiir;
+
+    if (ctx->fstate) {
+        for (ch = 0; ch < channels; ch++)
+            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_coeffsp(&ctx->fcoeffs);
+    if (ctx->fstate)
+        for (i = 0; i < ctx->avctx->channels; i++)
+            ff_iir_filter_free_statep(&ctx->fstate[i]);
+    av_freep(&ctx->fstate);
+    av_free(ctx);
+}