diff options
Diffstat (limited to 'libavcodec/psymodel.c')
| -rw-r--r-- | libavcodec/psymodel.c | 24 |
1 files changed, 16 insertions, 8 deletions
diff --git a/libavcodec/psymodel.c b/libavcodec/psymodel.c index a2af61175e..bfc85b3bc5 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 */ @@ -75,7 +75,7 @@ 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); @@ -88,6 +88,7 @@ typedef struct FFPsyPreprocessContext{ float stereo_att; struct FFIIRFilterCoeffs *fcoeffs; struct FFIIRFilterState **fstate; + struct FFIIRFilterContext fiir; }FFPsyPreprocessContext; #define FILT_ORDER 4 @@ -103,7 +104,10 @@ 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); @@ -112,6 +116,9 @@ 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; } @@ -119,11 +126,12 @@ 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); } } |