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-rw-r--r--libavcodec/psymodel.c38
1 files changed, 23 insertions, 15 deletions
diff --git a/libavcodec/psymodel.c b/libavcodec/psymodel.c
index 5179ede083..824eefb79e 100644
--- a/libavcodec/psymodel.c
+++ b/libavcodec/psymodel.c
@@ -2,20 +2,20 @@
* audio encoder psychoacoustic model
* Copyright (C) 2008 Konstantin Shishkov
*
- * This file is part of Libav.
+ * This file is part of FFmpeg.
*
- * Libav is free software; you can redistribute it and/or
+ * 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.
*
- * Libav is distributed in the hope that it will be useful,
+ * 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 Libav; if not, write to the Free Software
+ * License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
@@ -35,10 +35,10 @@ av_cold int ff_psy_init(FFPsyContext *ctx, AVCodecContext *avctx, int num_lens,
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);
+ 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);
if (!ctx->ch || !ctx->group || !ctx->bands || !ctx->num_bands) {
ff_psy_end(ctx);
@@ -81,7 +81,7 @@ FFPsyChannelGroup *ff_psy_find_group(FFPsyContext *ctx, int channel)
av_cold void ff_psy_end(FFPsyContext *ctx)
{
- if (ctx->model->end)
+ if (ctx->model && ctx->model->end)
ctx->model->end(ctx);
av_freep(&ctx->bands);
av_freep(&ctx->num_bands);
@@ -94,6 +94,7 @@ typedef struct FFPsyPreprocessContext{
float stereo_att;
struct FFIIRFilterCoeffs *fcoeffs;
struct FFIIRFilterState **fstate;
+ struct FFIIRFilterContext fiir;
}FFPsyPreprocessContext;
#define FILT_ORDER 4
@@ -111,12 +112,15 @@ av_cold struct FFPsyPreprocessContext* ff_psy_preprocess_init(AVCodecContext *av
if (avctx->cutoff > 0)
cutoff_coeff = 2.0 * avctx->cutoff / avctx->sample_rate;
- if (cutoff_coeff)
+ if (!cutoff_coeff && avctx->codec_id == AV_CODEC_ID_AAC)
+ cutoff_coeff = 2.0 * AAC_CUTOFF(avctx) / 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(sizeof(ctx->fstate[0]) * avctx->channels);
+ ctx->fstate = av_mallocz_array(sizeof(ctx->fstate[0]), avctx->channels);
if (!ctx->fstate) {
av_free(ctx);
return NULL;
@@ -124,6 +128,9 @@ av_cold struct FFPsyPreprocessContext* ff_psy_preprocess_init(AVCodecContext *av
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;
}
@@ -131,21 +138,22 @@ void ff_psy_preprocess(struct FFPsyPreprocessContext *ctx, float **audio, int ch
{
int ch;
int frame_size = ctx->avctx->frame_size;
+ FFIIRFilterContext *iir = &ctx->fiir;
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);
+ 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);
+ ff_iir_filter_free_coeffsp(&ctx->fcoeffs);
if (ctx->fstate)
for (i = 0; i < ctx->avctx->channels; i++)
- ff_iir_filter_free_state(ctx->fstate[i]);
+ ff_iir_filter_free_statep(&ctx->fstate[i]);
av_freep(&ctx->fstate);
av_free(ctx);
}