diff resample.c @ 0:986e461dc072 libavcodec

Initial revision

--- /dev/null	Thu Jan 01 00:00:00 1970 +0000
+++ b/resample.c	Sun Jul 22 14:18:56 2001 +0000
@@ -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 <stdlib.h>
+#include <stdio.h>
+#include <string.h>
+#include <math.h>
+#include "avcodec.h"
+
+#define NDEBUG
+#include <assert.h>
+
+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;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;
+}
+
+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;i<s->filter_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;i<s->filter_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);
+}