* audio encoder psychoacoustic model
* Copyright (C) 2008 Konstantin Shishkov
*
- * This file is part of FFmpeg.
+ * This file is part of Libav.
*
- * FFmpeg is free software; you can redistribute it and/or
+ * Libav 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,
+ * Libav 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
+ * License along with Libav; 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)
+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->psy_bands = av_mallocz(sizeof(FFPsyBand) * PSY_MAX_BANDS * avctx->channels);
+ 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);
+
+ 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 CODEC_ID_AAC:
+ case AV_CODEC_ID_AAC:
ctx->model = &ff_aac_psy_model;
break;
}
return 0;
}
-FFPsyWindowInfo ff_psy_suggest_window(FFPsyContext *ctx,
- const int16_t *audio, const int16_t *la,
- int channel, int prev_type)
+FFPsyChannelGroup *ff_psy_find_group(FFPsyContext *ctx, int channel)
{
- return ctx->model->window(ctx, audio, la, channel, prev_type);
-}
+ int i = 0, ch = 0;
-void ff_psy_set_band_info(FFPsyContext *ctx, int channel,
- const float *coeffs, const FFPsyWindowInfo *wi)
-{
- ctx->model->analyze(ctx, channel, coeffs, wi);
+ while (ch <= channel)
+ ch += ctx->group[i++].num_ch;
+
+ return &ctx->group[i-1];
}
av_cold void ff_psy_end(FFPsyContext *ctx)
ctx->model->end(ctx);
av_freep(&ctx->bands);
av_freep(&ctx->num_bands);
- av_freep(&ctx->psy_bands);
+ av_freep(&ctx->group);
+ av_freep(&ctx->ch);
}
typedef struct FFPsyPreprocessContext{
int i;
float cutoff_coeff = 0;
ctx = av_mallocz(sizeof(FFPsyPreprocessContext));
+ if (!ctx)
+ return NULL;
ctx->avctx = avctx;
if (avctx->cutoff > 0)
cutoff_coeff = 2.0 * avctx->cutoff / 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);
+ 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);
+ if (!ctx->fstate) {
+ av_free(ctx);
+ return NULL;
+ }
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,
- const int16_t *audio, int16_t *dest,
- int tag, int channels)
+void ff_psy_preprocess(struct FFPsyPreprocessContext *ctx, float **audio, int channels)
{
- int ch, i;
+ int ch;
+ int frame_size = ctx->avctx->frame_size;
+
if (ctx->fstate) {
for (ch = 0; ch < channels; ch++)
- ff_iir_filter(ctx->fcoeffs, ctx->fstate[tag+ch], ctx->avctx->frame_size,
- audio + ch, ctx->avctx->channels,
- dest + ch, ctx->avctx->channels);
- } else {
- for (ch = 0; ch < channels; ch++)
- for (i = 0; i < ctx->avctx->frame_size; i++)
- dest[i*ctx->avctx->channels + ch] = audio[i*ctx->avctx->channels + ch];
+ ff_iir_filter_flt(ctx->fcoeffs, ctx->fstate[ch], frame_size,
+ &audio[ch][frame_size], 1, &audio[ch][frame_size], 1);
}
}
av_freep(&ctx->fstate);
av_free(ctx);
}
-
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