so extend decoder to output only one channel for it. This fixes issue 2368. Originally committed as revision 25790 to svn://svn.ffmpeg.org/ffmpeg/trunk
		
			
				
	
	
		
			103 lines
		
	
	
		
			3.2 KiB
		
	
	
	
		
			C
		
	
	
	
	
	
			
		
		
	
	
			103 lines
		
	
	
		
			3.2 KiB
		
	
	
	
		
			C
		
	
	
	
	
	
/*
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 * Musepack decoder core
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 * Copyright (c) 2006 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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/**
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 * @file
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 * Musepack decoder core
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 * MPEG Audio Layer 1/2 -like codec with frames of 1152 samples
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 * divided into 32 subbands.
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 */
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#include "avcodec.h"
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#include "get_bits.h"
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#include "dsputil.h"
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#include "mpegaudio.h"
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#include "mpc.h"
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#include "mpcdata.h"
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void ff_mpc_init(void)
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{
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    ff_mpa_synth_init(ff_mpa_synth_window);
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}
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/**
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 * Process decoded Musepack data and produce PCM
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 */
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static void mpc_synth(MPCContext *c, int16_t *out, int channels)
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{
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    int dither_state = 0;
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    int i, ch;
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    OUT_INT samples[MPA_MAX_CHANNELS * MPA_FRAME_SIZE], *samples_ptr;
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    for(ch = 0;  ch < channels; ch++){
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        samples_ptr = samples + ch;
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        for(i = 0; i < SAMPLES_PER_BAND; i++) {
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            ff_mpa_synth_filter(c->synth_buf[ch], &(c->synth_buf_offset[ch]),
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                                ff_mpa_synth_window, &dither_state,
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                                samples_ptr, channels,
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                                c->sb_samples[ch][i]);
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            samples_ptr += 32 * channels;
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        }
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    }
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    for(i = 0; i < MPC_FRAME_SIZE*channels; i++)
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        *out++=samples[i];
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}
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void ff_mpc_dequantize_and_synth(MPCContext * c, int maxband, void *data, int channels)
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{
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    int i, j, ch;
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    Band *bands = c->bands;
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    int off;
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    float mul;
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    /* dequantize */
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    memset(c->sb_samples, 0, sizeof(c->sb_samples));
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    off = 0;
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    for(i = 0; i <= maxband; i++, off += SAMPLES_PER_BAND){
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        for(ch = 0; ch < 2; ch++){
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            if(bands[i].res[ch]){
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                j = 0;
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                mul = mpc_CC[bands[i].res[ch]] * mpc_SCF[bands[i].scf_idx[ch][0]];
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                for(; j < 12; j++)
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                    c->sb_samples[ch][j][i] = mul * c->Q[ch][j + off];
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                mul = mpc_CC[bands[i].res[ch]] * mpc_SCF[bands[i].scf_idx[ch][1]];
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                for(; j < 24; j++)
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                    c->sb_samples[ch][j][i] = mul * c->Q[ch][j + off];
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                mul = mpc_CC[bands[i].res[ch]] * mpc_SCF[bands[i].scf_idx[ch][2]];
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                for(; j < 36; j++)
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                    c->sb_samples[ch][j][i] = mul * c->Q[ch][j + off];
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            }
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        }
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        if(bands[i].msf){
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            int t1, t2;
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            for(j = 0; j < SAMPLES_PER_BAND; j++){
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                t1 = c->sb_samples[0][j][i];
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                t2 = c->sb_samples[1][j][i];
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                c->sb_samples[0][j][i] = t1 + t2;
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                c->sb_samples[1][j][i] = t1 - t2;
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            }
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        }
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    }
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    mpc_synth(c, data, channels);
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}
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