mirror of https://github.com/FFmpeg/FFmpeg.git
Originally committed as revision 19375 to svn://svn.ffmpeg.org/ffmpeg/trunkrelease/0.6
Alex Converse
16 years ago
parent
5e039e1b4c
commit
78e65cd772
10 changed files with 1992 additions and 115 deletions
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/*
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* AAC encoder |
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* Copyright (C) 2008 Konstantin Shishkov |
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* |
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* This file is part of FFmpeg. |
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* |
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* FFmpeg is free software; you can redistribute it and/or |
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* modify it under the terms of the GNU Lesser General Public |
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* License as published by the Free Software Foundation; either |
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* version 2.1 of the License, or (at your option) any later version. |
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* |
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* FFmpeg is distributed in the hope that it will be useful, |
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* but WITHOUT ANY WARRANTY; without even the implied warranty of |
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU |
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* Lesser General Public License for more details. |
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* |
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* You should have received a copy of the GNU Lesser General Public |
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* License along with FFmpeg; if not, write to the Free Software |
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA |
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*/ |
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#ifndef AVCODEC_AACENC_H |
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#define AVCODEC_AACENC_H |
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#include "avcodec.h" |
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#include "put_bits.h" |
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#include "dsputil.h" |
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#include "aac.h" |
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#include "psymodel.h" |
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struct AACEncContext; |
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typedef struct AACCoefficientsEncoder{ |
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void (*search_for_quantizers)(AVCodecContext *avctx, struct AACEncContext *s, |
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SingleChannelElement *sce, const float lambda); |
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void (*encode_window_bands_info)(struct AACEncContext *s, SingleChannelElement *sce, |
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int win, int group_len, const float lambda); |
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void (*quantize_and_encode_band)(struct AACEncContext *s, PutBitContext *pb, const float *in, int size, |
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int scale_idx, int cb, const float lambda); |
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void (*search_for_ms)(struct AACEncContext *s, ChannelElement *cpe, const float lambda); |
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}AACCoefficientsEncoder; |
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extern AACCoefficientsEncoder ff_aac_coders[]; |
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/**
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* AAC encoder context |
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*/ |
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typedef struct AACEncContext { |
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PutBitContext pb; |
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MDCTContext mdct1024; ///< long (1024 samples) frame transform context
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MDCTContext mdct128; ///< short (128 samples) frame transform context
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DSPContext dsp; |
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DECLARE_ALIGNED_16(FFTSample, output[2048]); ///< temporary buffer for MDCT input coefficients
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int16_t* samples; ///< saved preprocessed input
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int samplerate_index; ///< MPEG-4 samplerate index
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ChannelElement *cpe; ///< channel elements
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FFPsyContext psy; |
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struct FFPsyPreprocessContext* psypp; |
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AACCoefficientsEncoder *coder; |
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int cur_channel; |
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int last_frame; |
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float lambda; |
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DECLARE_ALIGNED_16(int, qcoefs[96][2]); ///< quantized coefficients
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DECLARE_ALIGNED_16(float, scoefs[1024]); ///< scaled coefficients
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} AACEncContext; |
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#endif /* AVCODEC_AACENC_H */ |
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/*
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* audio encoder psychoacoustic model |
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* Copyright (C) 2008 Konstantin Shishkov |
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* |
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* This file is part of FFmpeg. |
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* |
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* FFmpeg is free software; you can redistribute it and/or |
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* modify it under the terms of the GNU Lesser General Public |
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* License as published by the Free Software Foundation; either |
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* version 2.1 of the License, or (at your option) any later version. |
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* |
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* FFmpeg is distributed in the hope that it will be useful, |
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* but WITHOUT ANY WARRANTY; without even the implied warranty of |
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU |
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* Lesser General Public License for more details. |
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* |
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* You should have received a copy of the GNU Lesser General Public |
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* License along with FFmpeg; if not, write to the Free Software |
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA |
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*/ |
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#include "avcodec.h" |
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#include "psymodel.h" |
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#include "iirfilter.h" |
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extern const FFPsyModel ff_aac_psy_model; |
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av_cold int ff_psy_init(FFPsyContext *ctx, AVCodecContext *avctx, |
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int num_lens, |
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const uint8_t **bands, const int* num_bands) |
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{ |
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ctx->avctx = avctx; |
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ctx->psy_bands = av_mallocz(sizeof(FFPsyBand) * PSY_MAX_BANDS * avctx->channels); |
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ctx->bands = av_malloc (sizeof(ctx->bands[0]) * num_lens); |
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ctx->num_bands = av_malloc (sizeof(ctx->num_bands[0]) * num_lens); |
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memcpy(ctx->bands, bands, sizeof(ctx->bands[0]) * num_lens); |
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memcpy(ctx->num_bands, num_bands, sizeof(ctx->num_bands[0]) * num_lens); |
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switch(ctx->avctx->codec_id){ |
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case CODEC_ID_AAC: |
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ctx->model = &ff_aac_psy_model; |
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break; |
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} |
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if(ctx->model->init) |
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return ctx->model->init(ctx); |
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return 0; |
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} |
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FFPsyWindowInfo ff_psy_suggest_window(FFPsyContext *ctx, |
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const int16_t *audio, const int16_t *la, |
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int channel, int prev_type) |
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{ |
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return ctx->model->window(ctx, audio, la, channel, prev_type); |
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} |
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void ff_psy_set_band_info(FFPsyContext *ctx, int channel, |
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const float *coeffs, FFPsyWindowInfo *wi) |
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{ |
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ctx->model->analyze(ctx, channel, coeffs, wi); |
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} |
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av_cold void ff_psy_end(FFPsyContext *ctx) |
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{ |
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if(ctx->model->end) |
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ctx->model->end(ctx); |
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av_freep(&ctx->bands); |
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av_freep(&ctx->num_bands); |
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av_freep(&ctx->psy_bands); |
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} |
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typedef struct FFPsyPreprocessContext{ |
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AVCodecContext *avctx; |
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float stereo_att; |
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struct FFIIRFilterCoeffs *fcoeffs; |
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struct FFIIRFilterState **fstate; |
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}FFPsyPreprocessContext; |
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#define FILT_ORDER 4 |
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av_cold struct FFPsyPreprocessContext* ff_psy_preprocess_init(AVCodecContext *avctx) |
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{ |
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FFPsyPreprocessContext *ctx; |
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int i; |
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float cutoff_coeff; |
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ctx = av_mallocz(sizeof(FFPsyPreprocessContext)); |
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ctx->avctx = avctx; |
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if(avctx->flags & CODEC_FLAG_QSCALE) |
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cutoff_coeff = 1.0f / av_clip(1 + avctx->global_quality / FF_QUALITY_SCALE, 1, 8); |
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else |
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cutoff_coeff = avctx->bit_rate / (4.0f * avctx->sample_rate * avctx->channels); |
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ctx->fcoeffs = ff_iir_filter_init_coeffs(FF_FILTER_TYPE_BUTTERWORTH, FF_FILTER_MODE_LOWPASS, |
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FILT_ORDER, cutoff_coeff, 0.0, 0.0); |
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if(ctx->fcoeffs){ |
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ctx->fstate = av_mallocz(sizeof(ctx->fstate[0]) * avctx->channels); |
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for(i = 0; i < avctx->channels; i++) |
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ctx->fstate[i] = ff_iir_filter_init_state(FILT_ORDER); |
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} |
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return ctx; |
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} |
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void ff_psy_preprocess(struct FFPsyPreprocessContext *ctx, |
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const int16_t *audio, int16_t *dest, |
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int tag, int channels) |
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{ |
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int ch, i; |
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if(ctx->fstate){ |
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for(ch = 0; ch < channels; ch++){ |
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ff_iir_filter(ctx->fcoeffs, ctx->fstate[tag+ch], ctx->avctx->frame_size, |
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audio + ch, ctx->avctx->channels, |
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dest + ch, ctx->avctx->channels); |
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} |
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}else{ |
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for(ch = 0; ch < channels; ch++){ |
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for(i = 0; i < ctx->avctx->frame_size; i++) |
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dest[i*ctx->avctx->channels + ch] = audio[i*ctx->avctx->channels + ch]; |
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} |
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} |
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} |
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av_cold void ff_psy_preprocess_end(struct FFPsyPreprocessContext *ctx) |
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{ |
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int i; |
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ff_iir_filter_free_coeffs(ctx->fcoeffs); |
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if (ctx->fstate) |
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for (i = 0; i < ctx->avctx->channels; i++) |
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ff_iir_filter_free_state(ctx->fstate[i]); |
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av_freep(&ctx->fstate); |
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} |
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@ -0,0 +1,158 @@ |
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/*
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* audio encoder psychoacoustic model |
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* Copyright (C) 2008 Konstantin Shishkov |
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* |
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* This file is part of FFmpeg. |
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* |
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* FFmpeg is free software; you can redistribute it and/or |
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* modify it under the terms of the GNU Lesser General Public |
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* License as published by the Free Software Foundation; either |
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* version 2.1 of the License, or (at your option) any later version. |
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* |
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* FFmpeg is distributed in the hope that it will be useful, |
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* but WITHOUT ANY WARRANTY; without even the implied warranty of |
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU |
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* Lesser General Public License for more details. |
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* |
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* You should have received a copy of the GNU Lesser General Public |
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* License along with FFmpeg; if not, write to the Free Software |
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA |
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*/ |
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#ifndef AVCODEC_PSYMODEL_H |
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#define AVCODEC_PSYMODEL_H |
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#include "avcodec.h" |
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/** maximum possible number of bands */ |
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#define PSY_MAX_BANDS 128 |
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/**
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* single band psychoacoustic information |
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*/ |
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typedef struct FFPsyBand{ |
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int bits; |
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float energy; |
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float threshold; |
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float distortion; |
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float perceptual_weight; |
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}FFPsyBand; |
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/**
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* windowing related information |
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*/ |
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typedef struct FFPsyWindowInfo{ |
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int window_type[3]; ///< window type (short/long/transitional, etc.) - current, previous and next
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int window_shape; ///< window shape (sine/KBD/whatever)
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int num_windows; ///< number of windows in a frame
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int grouping[8]; ///< window grouping (for e.g. AAC)
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int *window_sizes; ///< sequence of window sizes inside one frame (for eg. WMA)
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}FFPsyWindowInfo; |
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/**
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* context used by psychoacoustic model |
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*/ |
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typedef struct FFPsyContext{ |
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AVCodecContext *avctx; ///< encoder context
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const struct FFPsyModel *model; ///< encoder-specific model functions
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FFPsyBand *psy_bands; ///< frame bands information
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uint8_t **bands; ///< scalefactor band sizes for possible frame sizes
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int *num_bands; ///< number of scalefactor bands for possible frame sizes
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int num_lens; ///< number of scalefactor band sets
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void* model_priv_data; ///< psychoacoustic model implementation private data
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}FFPsyContext; |
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/**
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* codec-specific psychoacoustic model implementation |
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*/ |
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typedef struct FFPsyModel { |
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const char *name; |
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int (*init) (FFPsyContext *apc); |
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FFPsyWindowInfo (*window)(FFPsyContext *ctx, const int16_t *audio, const int16_t *la, int channel, int prev_type); |
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void (*analyze)(FFPsyContext *ctx, int channel, const float *coeffs, FFPsyWindowInfo *wi); |
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void (*end) (FFPsyContext *apc); |
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}FFPsyModel; |
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/**
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* Initialize psychoacoustic model. |
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* |
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* @param ctx model context |
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* @param avctx codec context |
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* @param num_lens number of possible frame lengths |
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* @param bands scalefactor band lengths for all frame lengths |
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* @param num_bands number of scalefactor bands for all frame lengths |
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* |
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* @return zero if successful, a negative value if not |
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*/ |
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av_cold int ff_psy_init(FFPsyContext *ctx, AVCodecContext *avctx, |
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int num_lens, |
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const uint8_t **bands, const int* num_bands); |
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/**
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* Suggest window sequence for channel. |
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* |
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* @param ctx model context |
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* @param audio samples for the current frame |
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* @param la lookahead samples (NULL when unavailable) |
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* @param channel number of channel element to analyze |
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* @param prev_type previous window type |
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* |
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* @return suggested window information in a structure |
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*/ |
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FFPsyWindowInfo ff_psy_suggest_window(FFPsyContext *ctx, |
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const int16_t *audio, const int16_t *la, |
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int channel, int prev_type); |
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/**
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* Perform psychoacoustic analysis and set band info (threshold, energy). |
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* |
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* @param ctx model context |
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* @param channel audio channel number |
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* @param coeffs pointer to the transformed coefficients |
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* @param wi window information |
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*/ |
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void ff_psy_set_band_info(FFPsyContext *ctx, int channel, const float *coeffs, |
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FFPsyWindowInfo *wi); |
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/**
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* Cleanup model context at the end. |
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* |
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* @param ctx model context |
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*/ |
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av_cold void ff_psy_end(FFPsyContext *ctx); |
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/**************************************************************************
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* Audio preprocessing stuff. * |
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* This should be moved into some audio filter eventually. * |
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**************************************************************************/ |
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struct FFPsyPreprocessContext; |
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/**
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* psychoacoustic model audio preprocessing initialization |
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*/ |
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av_cold struct FFPsyPreprocessContext* ff_psy_preprocess_init(AVCodecContext *avctx); |
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/**
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* Preprocess several channel in audio frame in order to compress it better. |
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* |
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* @param ctx preprocessing context |
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* @param audio samples to preprocess |
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* @param dest place to put filtered samples |
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* @param tag channel number |
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* @param channels number of channel to preprocess (some additional work may be done on stereo pair) |
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*/ |
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void ff_psy_preprocess(struct FFPsyPreprocessContext *ctx, |
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const int16_t *audio, int16_t *dest, |
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int tag, int channels); |
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/**
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* Cleanup audio preprocessing module. |
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*/ |
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av_cold void ff_psy_preprocess_end(struct FFPsyPreprocessContext *ctx); |
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#endif /* AVCODEC_PSYMODEL_H */ |
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