diff src/ffmpeg/libavcodec/resample.c @ 808:e8776388b02a trunk

[svn] - add ffmpeg

--- /dev/null	Thu Jan 01 00:00:00 1970 +0000
+++ b/src/ffmpeg/libavcodec/resample.c	Mon Mar 12 11:18:54 2007 -0700
@@ -0,0 +1,249 @@
+/*
+ * Sample rate convertion for both audio and video
+ * Copyright (c) 2000 Fabrice Bellard.
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg 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
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+/**
+ * @file resample.c
+ * Sample rate convertion for both audio and video.
+ */
+
+#include "avcodec.h"
+
+struct AVResampleContext;
+
+struct ReSampleContext {
+    struct AVResampleContext *resample_context;
+    short *temp[2];
+    int temp_len;
+    float ratio;
+    /* channel convert */
+    int input_channels, output_channels, filter_channels;
+};
+
+/* 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 void ac3_5p1_mux(short *output, short *input1, short *input2, int n)
+{
+    int i;
+    short l,r;
+
+    for(i=0;i<n;i++) {
+      l=*input1++;
+      r=*input2++;
+      *output++ = l;           /* left */
+      *output++ = (l/2)+(r/2); /* center */
+      *output++ = r;           /* right */
+      *output++ = 0;           /* left surround */
+      *output++ = 0;           /* right surroud */
+      *output++ = 0;           /* low freq */
+    }
+}
+
+ReSampleContext *audio_resample_init(int output_channels, int input_channels,
+                                      int output_rate, int input_rate)
+{
+    ReSampleContext *s;
+
+    if ( input_channels > 2)
+      {
+        av_log(NULL, AV_LOG_ERROR, "Resampling with input channels greater than 2 unsupported.");
+        return NULL;
+      }
+
+    s = av_mallocz(sizeof(ReSampleContext));
+    if (!s)
+      {
+        av_log(NULL, AV_LOG_ERROR, "Can't allocate memory for resample context.");
+        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;
+
+/*
+ * ac3 output is the only case where filter_channels could be greater than 2.
+ * input channels can't be greater than 2, so resample the 2 channels and then
+ * expand to 6 channels after the resampling.
+ */
+    if(s->filter_channels>2)
+      s->filter_channels = 2;
+
+    s->resample_context= av_resample_init(output_rate, input_rate, 16, 10, 0, 1.0);
+
+    return s;
+}
+
+/* resample audio. 'nb_samples' is the number of input samples */
+/* XXX: optimize it ! */
+int audio_resample(ReSampleContext *s, short *output, short *input, int nb_samples)
+{
+    int i, nb_samples1;
+    short *bufin[2];
+    short *bufout[2];
+    short *buftmp2[2], *buftmp3[2];
+    int lenout;
+
+    if (s->input_channels == s->output_channels && s->ratio == 1.0 && 0) {
+        /* nothing to do */
+        memcpy(output, input, nb_samples * s->input_channels * sizeof(short));
+        return nb_samples;
+    }
+
+    /* XXX: move those malloc to resample init code */
+    for(i=0; i<s->filter_channels; i++){
+        bufin[i]= (short*) av_malloc( (nb_samples + s->temp_len) * sizeof(short) );
+        memcpy(bufin[i], s->temp[i], s->temp_len * sizeof(short));
+        buftmp2[i] = bufin[i] + s->temp_len;
+    }
+
+    /* make some zoom to avoid round pb */
+    lenout= (int)(nb_samples * s->ratio) + 16;
+    bufout[0]= (short*) av_malloc( lenout * sizeof(short) );
+    bufout[1]= (short*) av_malloc( lenout * sizeof(short) );
+
+    if (s->input_channels == 2 &&
+        s->output_channels == 1) {
+        buftmp3[0] = output;
+        stereo_to_mono(buftmp2[0], input, nb_samples);
+    } else if (s->output_channels >= 2 && s->input_channels == 1) {
+        buftmp3[0] = bufout[0];
+        memcpy(buftmp2[0], input, nb_samples*sizeof(short));
+    } else if (s->output_channels >= 2) {
+        buftmp3[0] = bufout[0];
+        buftmp3[1] = bufout[1];
+        stereo_split(buftmp2[0], buftmp2[1], input, nb_samples);
+    } else {
+        buftmp3[0] = output;
+        memcpy(buftmp2[0], input, nb_samples*sizeof(short));
+    }
+
+    nb_samples += s->temp_len;
+
+    /* resample each channel */
+    nb_samples1 = 0; /* avoid warning */
+    for(i=0;i<s->filter_channels;i++) {
+        int consumed;
+        int is_last= i+1 == s->filter_channels;
+
+        nb_samples1 = av_resample(s->resample_context, buftmp3[i], bufin[i], &consumed, nb_samples, lenout, is_last);
+        s->temp_len= nb_samples - consumed;
+        s->temp[i]= av_realloc(s->temp[i], s->temp_len*sizeof(short));
+        memcpy(s->temp[i], bufin[i] + consumed, s->temp_len*sizeof(short));
+    }
+
+    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);
+    } else if (s->output_channels == 6) {
+        ac3_5p1_mux(output, buftmp3[0], buftmp3[1], nb_samples1);
+    }
+
+    for(i=0; i<s->filter_channels; i++)
+        av_free(bufin[i]);
+
+    av_free(bufout[0]);
+    av_free(bufout[1]);
+    return nb_samples1;
+}
+
+void audio_resample_close(ReSampleContext *s)
+{
+    av_resample_close(s->resample_context);
+    av_freep(&s->temp[0]);
+    av_freep(&s->temp[1]);
+    av_free(s);
+}