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Diffstat (limited to 'contrib/ffmpeg/libavcodec/resample.c')
-rw-r--r--contrib/ffmpeg/libavcodec/resample.c250
1 files changed, 0 insertions, 250 deletions
diff --git a/contrib/ffmpeg/libavcodec/resample.c b/contrib/ffmpeg/libavcodec/resample.c
deleted file mode 100644
index ea5c6d61c..000000000
--- a/contrib/ffmpeg/libavcodec/resample.c
+++ /dev/null
@@ -1,250 +0,0 @@
-/*
- * 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;
-
-#define TAPS 16
- s->resample_context= av_resample_init(output_rate, input_rate, TAPS, 10, 0, 0.8);
-
- 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);
-}
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