From de6d9b6404bfd1c589799142da5a95428f146edd Mon Sep 17 00:00:00 2001 From: Fabrice Bellard Date: Sun, 22 Jul 2001 14:18:56 +0000 Subject: Initial revision Originally committed as revision 5 to svn://svn.ffmpeg.org/ffmpeg/trunk --- libavcodec/resample.c | 301 ++++++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 301 insertions(+) create mode 100644 libavcodec/resample.c (limited to 'libavcodec/resample.c') diff --git a/libavcodec/resample.c b/libavcodec/resample.c new file mode 100644 index 0000000000..29445964f6 --- /dev/null +++ b/libavcodec/resample.c @@ -0,0 +1,301 @@ +/* + * 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 +#include +#include +#include +#include "avcodec.h" + +#define NDEBUG +#include + +typedef struct { + /* fractional resampling */ + UINT32 incr; /* fractional increment */ + UINT32 frac; + int last_sample; + /* integer down sample */ + int iratio; /* integer divison ratio */ + int icount, isum; + int inv; +} ReSampleChannelContext; + +struct ReSampleContext { + ReSampleChannelContext channel_ctx[2]; + float ratio; + /* channel convert */ + int input_channels, output_channels, filter_channels; +}; + + +#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--; + } +} + +/* n1: number of samples */ +static void mono_to_stereo(short *output, short *input, int n1) +{ + short *p, *q; + int n = n1; + int v; + + p = input; + q = output; + while (n >= 4) { + v = p[0]; q[0] = v; q[1] = v; + v = p[1]; q[2] = v; q[3] = v; + v = p[2]; q[4] = v; q[5] = v; + v = p[3]; q[6] = v; q[7] = v; + q += 8; + p += 4; + n -= 4; + } + while (n > 0) { + v = p[0]; q[0] = v; q[1] = v; + q += 2; + p += 1; + 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;iiratio > 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; +} + +ReSampleContext *audio_resample_init(int output_channels, int input_channels, + int output_rate, int input_rate) +{ + ReSampleContext *s; + int i; + + if (output_channels > 2 || input_channels > 2) + return NULL; + + s = av_mallocz(sizeof(ReSampleContext)); + if (!s) + return NULL; + + s->ratio = (float)output_rate / (float)input_rate; + + s->input_channels = input_channels; + s->output_channels = output_channels; + + s->filter_channels = s->input_channels; + if (s->output_channels < s->filter_channels) + s->filter_channels = s->output_channels; + + for(i=0;ifilter_channels;i++) { + init_mono_resample(&s->channel_ctx[i], s->ratio); + } + return s; +} + +/* 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 bufin[2][nb_samples]; + short bufout[2][(int)(nb_samples * s->ratio) + 16]; /* make some zoom to avoid round pb */ + short *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) { + buftmp2[0] = bufin[0]; + buftmp3[0] = output; + stereo_to_mono(buftmp2[0], input, nb_samples); + } else if (s->output_channels == 2 && s->input_channels == 1) { + buftmp2[0] = input; + buftmp3[0] = bufout[0]; + } else if (s->output_channels == 2) { + buftmp2[0] = bufin[0]; + buftmp2[1] = bufin[1]; + buftmp3[0] = bufout[0]; + buftmp3[1] = bufout[1]; + stereo_split(buftmp2[0], buftmp2[1], input, nb_samples); + } else { + buftmp2[0] = input; + buftmp3[0] = output; + } + + /* resample each channel */ + nb_samples1 = 0; /* avoid warning */ + for(i=0;ifilter_channels;i++) { + nb_samples1 = mono_resample(&s->channel_ctx[i], buftmp3[i], buftmp2[i], nb_samples); + } + + if (s->output_channels == 2 && s->input_channels == 1) { + mono_to_stereo(output, buftmp3[0], nb_samples1); + } else if (s->output_channels == 2) { + stereo_mux(output, buftmp3[0], buftmp3[1], nb_samples1); + } + + return nb_samples1; +} + +void audio_resample_close(ReSampleContext *s) +{ + free(s); +} -- cgit v1.2.3