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* Sample rate convertion for both audio and video
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* Copyright (c) 2000 Gerard Lantau.
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* This program is free software; you can redistribute it and/or modify
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* it under the terms of the GNU General Public License as published by
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* the Free Software Foundation; either version 2 of the License, or
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* (at your option) any later version.
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* This program 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
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* GNU General Public License for more details.
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* You should have received a copy of the GNU General Public License
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* along with this program; if not, write to the Free Software
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* Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
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/* fractional resampling */
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UINT32 incr; /* fractional increment */
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/* integer down sample */
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int iratio; /* integer divison ratio */
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} ReSampleChannelContext;
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struct ReSampleContext {
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ReSampleChannelContext channel_ctx[2];
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int input_channels, output_channels, filter_channels;
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#define FRAC (1 << FRAC_BITS)
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static void init_mono_resample(ReSampleChannelContext *s, float ratio)
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s->iratio = (int)floor(ratio);
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s->incr = (int)((ratio / s->iratio) * FRAC);
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s->icount = s->iratio;
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s->inv = (FRAC / s->iratio);
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/* fractional audio resampling */
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static int fractional_resample(ReSampleChannelContext *s, short *output, short *input, int nb_samples)
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unsigned int frac, incr;
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pend = input + nb_samples;
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*q++ = (l0 * (FRAC - frac) + l1 * frac) >> FRAC_BITS;
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frac = frac + s->incr;
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while (frac >= FRAC) {
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static int integer_downsample(ReSampleChannelContext *s, short *output, short *input, int nb_samples)
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pend = input + nb_samples;
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*q++ = (sum * s->inv) >> FRAC_BITS;
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/* n1: number of samples */
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static void stereo_to_mono(short *output, short *input, int n1)
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q[0] = (p[0] + p[1]) >> 1;
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q[1] = (p[2] + p[3]) >> 1;
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q[2] = (p[4] + p[5]) >> 1;
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q[3] = (p[6] + p[7]) >> 1;
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q[0] = (p[0] + p[1]) >> 1;
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/* n1: number of samples */
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static void mono_to_stereo(short *output, short *input, int n1)
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v = p[0]; q[0] = v; q[1] = v;
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v = p[1]; q[2] = v; q[3] = v;
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v = p[2]; q[4] = v; q[5] = v;
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v = p[3]; q[6] = v; q[7] = v;
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v = p[0]; q[0] = v; q[1] = v;
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/* XXX: should use more abstract 'N' channels system */
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static void stereo_split(short *output1, short *output2, short *input, int n)
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*output1++ = *input++;
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*output2++ = *input++;
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static void stereo_mux(short *output, short *input1, short *input2, int n)
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*output++ = *input1++;
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*output++ = *input2++;
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static int mono_resample(ReSampleChannelContext *s, short *output, short *input, int nb_samples)
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buf1= (short*) malloc( nb_samples * sizeof(short) );
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/* first downsample by an integer factor with averaging filter */
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nb_samples = integer_downsample(s, buftmp, input, nb_samples);
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/* then do a fractional resampling with linear interpolation */
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if (s->incr != FRAC) {
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nb_samples = fractional_resample(s, output, buftmp, nb_samples);
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memcpy(output, buftmp, nb_samples * sizeof(short));
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ReSampleContext *audio_resample_init(int output_channels, int input_channels,
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int output_rate, int input_rate)
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if (output_channels > 2 || input_channels > 2)
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s = av_mallocz(sizeof(ReSampleContext));
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s->ratio = (float)output_rate / (float)input_rate;
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s->input_channels = input_channels;
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s->output_channels = output_channels;
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s->filter_channels = s->input_channels;
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if (s->output_channels < s->filter_channels)
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s->filter_channels = s->output_channels;
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for(i=0;i<s->filter_channels;i++) {
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init_mono_resample(&s->channel_ctx[i], s->ratio);
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/* resample audio. 'nb_samples' is the number of input samples */
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/* XXX: optimize it ! */
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/* XXX: do it with polyphase filters, since the quality here is
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HORRIBLE. Return the number of samples available in output */
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int audio_resample(ReSampleContext *s, short *output, short *input, int nb_samples)
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short *buftmp2[2], *buftmp3[2];
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if (s->input_channels == s->output_channels && s->ratio == 1.0) {
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memcpy(output, input, nb_samples * s->input_channels * sizeof(short));
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/* XXX: move those malloc to resample init code */
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bufin[0]= (short*) malloc( nb_samples * sizeof(short) );
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bufin[1]= (short*) malloc( nb_samples * sizeof(short) );
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/* make some zoom to avoid round pb */
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lenout= (int)(nb_samples * s->ratio) + 16;
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bufout[0]= (short*) malloc( lenout * sizeof(short) );
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bufout[1]= (short*) malloc( lenout * sizeof(short) );
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if (s->input_channels == 2 &&
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s->output_channels == 1) {
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buftmp2[0] = bufin[0];
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stereo_to_mono(buftmp2[0], input, nb_samples);
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} else if (s->output_channels == 2 && s->input_channels == 1) {
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buftmp3[0] = bufout[0];
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} else if (s->output_channels == 2) {
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buftmp2[0] = bufin[0];
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buftmp2[1] = bufin[1];
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buftmp3[0] = bufout[0];
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buftmp3[1] = bufout[1];
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stereo_split(buftmp2[0], buftmp2[1], input, nb_samples);
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/* resample each channel */
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nb_samples1 = 0; /* avoid warning */
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for(i=0;i<s->filter_channels;i++) {
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nb_samples1 = mono_resample(&s->channel_ctx[i], buftmp3[i], buftmp2[i], nb_samples);
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if (s->output_channels == 2 && s->input_channels == 1) {
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mono_to_stereo(output, buftmp3[0], nb_samples1);
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} else if (s->output_channels == 2) {
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stereo_mux(output, buftmp3[0], buftmp3[1], nb_samples1);
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void audio_resample_close(ReSampleContext *s)
added
removed
ffmpeg/libavcodec/resample.c
