diff options
Diffstat (limited to 'libav/resample.c')
| -rw-r--r-- | libav/resample.c | 245 |
1 files changed, 0 insertions, 245 deletions
diff --git a/libav/resample.c b/libav/resample.c deleted file mode 100644 index 008153b0d6..0000000000 --- a/libav/resample.c +++ /dev/null @@ -1,245 +0,0 @@ -/* - * Sample rate convertion for both audio and video - * Copyright (c) 2000 Gerard Lantau. - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program 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 General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, write to the Free Software - * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. - */ -#include <stdlib.h> -#include <stdio.h> -#include <string.h> -#include <netinet/in.h> -#include <math.h> -#include "avcodec.h" - -#define NDEBUG -#include <assert.h> - -#define FRAC_BITS 16 -#define FRAC (1 << FRAC_BITS) - -static void init_mono_resample(ReSampleChannelContext *s, float ratio) -{ - ratio = 1.0 / ratio; - s->iratio = (int)floor(ratio); - if (s->iratio == 0) - s->iratio = 1; - s->incr = (int)((ratio / s->iratio) * FRAC); - s->frac = 0; - s->last_sample = 0; - s->icount = s->iratio; - s->isum = 0; - s->inv = (FRAC / s->iratio); -} - -/* fractional audio resampling */ -static int fractional_resample(ReSampleChannelContext *s, short *output, short *input, int nb_samples) -{ - unsigned int frac, incr; - int l0, l1; - short *q, *p, *pend; - - l0 = s->last_sample; - incr = s->incr; - frac = s->frac; - - p = input; - pend = input + nb_samples; - q = output; - - l1 = *p++; - for(;;) { - /* interpolate */ - *q++ = (l0 * (FRAC - frac) + l1 * frac) >> FRAC_BITS; - frac = frac + s->incr; - while (frac >= FRAC) { - if (p >= pend) - goto the_end; - frac -= FRAC; - l0 = l1; - l1 = *p++; - } - } - the_end: - s->last_sample = l1; - s->frac = frac; - return q - output; -} - -static int integer_downsample(ReSampleChannelContext *s, short *output, short *input, int nb_samples) -{ - short *q, *p, *pend; - int c, sum; - - p = input; - pend = input + nb_samples; - q = output; - - c = s->icount; - sum = s->isum; - - for(;;) { - sum += *p++; - if (--c == 0) { - *q++ = (sum * s->inv) >> FRAC_BITS; - c = s->iratio; - sum = 0; - } - if (p >= pend) - break; - } - s->isum = sum; - s->icount = c; - return q - output; -} - -/* n1: number of samples */ -static void stereo_to_mono(short *output, short *input, int n1) -{ - short *p, *q; - int n = n1; - - p = input; - q = output; - while (n >= 4) { - q[0] = (p[0] + p[1]) >> 1; - q[1] = (p[2] + p[3]) >> 1; - q[2] = (p[4] + p[5]) >> 1; - q[3] = (p[6] + p[7]) >> 1; - q += 4; - p += 8; - n -= 4; - } - while (n > 0) { - q[0] = (p[0] + p[1]) >> 1; - q++; - p += 2; - n--; - } -} - -/* XXX: should use more abstract 'N' channels system */ -static void stereo_split(short *output1, short *output2, short *input, int n) -{ - int i; - - for(i=0;i<n;i++) { - *output1++ = *input++; - *output2++ = *input++; - } -} - -static void stereo_mux(short *output, short *input1, short *input2, int n) -{ - int i; - - for(i=0;i<n;i++) { - *output++ = *input1++; - *output++ = *input2++; - } -} - -static int mono_resample(ReSampleChannelContext *s, short *output, short *input, int nb_samples) -{ - short buf1[nb_samples]; - short *buftmp; - - /* first downsample by an integer factor with averaging filter */ - if (s->iratio > 1) { - buftmp = buf1; - nb_samples = integer_downsample(s, buftmp, input, nb_samples); - } else { - buftmp = input; - } - - /* then do a fractional resampling with linear interpolation */ - if (s->incr != FRAC) { - nb_samples = fractional_resample(s, output, buftmp, nb_samples); - } else { - memcpy(output, buftmp, nb_samples * sizeof(short)); - } - return nb_samples; -} - -/* ratio = output_rate / input_rate */ -int audio_resample_init(ReSampleContext *s, - int output_channels, int input_channels, - int output_rate, int input_rate) -{ - int i; - - s->ratio = (float)output_rate / (float)input_rate; - - if (output_channels > 2 || input_channels > 2) - return -1; - s->input_channels = input_channels; - s->output_channels = output_channels; - - for(i=0;i<output_channels;i++) { - init_mono_resample(&s->channel_ctx[i], s->ratio); - } - return 0; -} - -/* resample audio. 'nb_samples' is the number of input samples */ -/* XXX: optimize it ! */ -/* XXX: do it with polyphase filters, since the quality here is - HORRIBLE. Return the number of samples available in output */ -int audio_resample(ReSampleContext *s, short *output, short *input, int nb_samples) -{ - int i, nb_samples1; - short buf[5][nb_samples]; - short *buftmp1, *buftmp2[2], *buftmp3[2]; - - if (s->input_channels == s->output_channels && s->ratio == 1.0) { - /* nothing to do */ - memcpy(output, input, nb_samples * s->input_channels * sizeof(short)); - return nb_samples; - } - - if (s->input_channels == 2 && - s->output_channels == 1) { - buftmp1 = buf[0]; - stereo_to_mono(buftmp1, input, nb_samples); - } else if (s->input_channels == 1 && - s->output_channels == 2) { - /* XXX: do it */ - abort(); - } else { - buftmp1 = input; - } - - if (s->output_channels == 2) { - buftmp2[0] = buf[1]; - buftmp2[1] = buf[2]; - buftmp3[0] = buf[3]; - buftmp3[1] = buf[4]; - stereo_split(buftmp2[0], buftmp2[1], buftmp1, nb_samples); - } else { - buftmp2[0] = buftmp1; - buftmp3[0] = output; - } - - /* resample each channel */ - nb_samples1 = 0; /* avoid warning */ - for(i=0;i<s->output_channels;i++) { - nb_samples1 = mono_resample(&s->channel_ctx[i], buftmp3[i], buftmp2[i], nb_samples); - } - - if (s->output_channels == 2) { - stereo_mux(output, buftmp3[0], buftmp3[1], nb_samples1); - } - - return nb_samples1; -} |