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-rw-r--r--libavcodec/resample.c301
1 files changed, 301 insertions, 0 deletions
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 <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);
+}
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