lavc: remove disabled FF_API_AVCODEC_RESAMPLE cruft

1 files changed, 0 insertions, 379 deletions

diff --git a/libavcodec/resample.c b/libavcodec/resample.c
deleted file mode 100644
index 1b3bb834f3..0000000000
--- a/libavcodec/resample.c
+++ /dev/null
@@ -1,379 +0,0 @@
-/*
- * samplerate conversion for both audio and video
- * Copyright (c) 2000 Fabrice Bellard
- *
- * This file is part of Libav.
- *
- * Libav 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.
- *
- * Libav 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 Libav; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-/**
- * @file
- * samplerate conversion for both audio and video
- */
-
-#include <string.h>
-
-#include "avcodec.h"
-#include "audioconvert.h"
-#include "libavutil/opt.h"
-#include "libavutil/mem.h"
-#include "libavutil/samplefmt.h"
-
-#if FF_API_AVCODEC_RESAMPLE
-
-#define MAX_CHANNELS 8
-
-struct AVResampleContext;
-
-static const char *context_to_name(void *ptr)
-{
-    return "audioresample";
-}
-
-static const AVOption options[] = {{NULL}};
-static const AVClass audioresample_context_class = {
-    "ReSampleContext", context_to_name, options, LIBAVUTIL_VERSION_INT
-};
-
-struct ReSampleContext {
-    struct AVResampleContext *resample_context;
-    short *temp[MAX_CHANNELS];
-    int temp_len;
-    float ratio;
-    /* channel convert */
-    int input_channels, output_channels, filter_channels;
-    AVAudioConvert *convert_ctx[2];
-    enum AVSampleFormat sample_fmt[2]; ///< input and output sample format
-    unsigned sample_size[2];           ///< size of one sample in sample_fmt
-    short *buffer[2];                  ///< buffers used for conversion to S16
-    unsigned buffer_size[2];           ///< sizes of allocated buffers
-};
-
-/* 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--;
-    }
-}
-
-static void deinterleave(short **output, short *input, int channels, int samples)
-{
-    int i, j;
-
-    for (i = 0; i < samples; i++) {
-        for (j = 0; j < channels; j++) {
-            *output[j]++ = *input++;
-        }
-    }
-}
-
-static void interleave(short *output, short **input, int channels, int samples)
-{
-    int i, j;
-
-    for (i = 0; i < samples; i++) {
-        for (j = 0; j < channels; j++) {
-            *output++ = *input[j]++;
-        }
-    }
-}
-
-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 *av_audio_resample_init(int output_channels, int input_channels,
-                                        int output_rate, int input_rate,
-                                        enum AVSampleFormat sample_fmt_out,
-                                        enum AVSampleFormat sample_fmt_in,
-                                        int filter_length, int log2_phase_count,
-                                        int linear, double cutoff)
-{
-    ReSampleContext *s;
-
-    if (input_channels > MAX_CHANNELS) {
-        av_log(NULL, AV_LOG_ERROR,
-               "Resampling with input channels greater than %d is unsupported.\n",
-               MAX_CHANNELS);
-        return NULL;
-    }
-    if (output_channels != input_channels &&
-        (input_channels  > 2 ||
-         output_channels > 2 &&
-         !(output_channels == 6 && input_channels == 2))) {
-        av_log(NULL, AV_LOG_ERROR,
-               "Resampling output channel count must be 1 or 2 for mono input; 1, 2 or 6 for stereo input; or N for N channel input.\n");
-        return NULL;
-    }
-
-    s = av_mallocz(sizeof(ReSampleContext));
-    if (!s) {
-        av_log(NULL, AV_LOG_ERROR, "Can't allocate memory for resample context.\n");
-        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;
-
-    s->sample_fmt[0]  = sample_fmt_in;
-    s->sample_fmt[1]  = sample_fmt_out;
-    s->sample_size[0] = av_get_bytes_per_sample(s->sample_fmt[0]);
-    s->sample_size[1] = av_get_bytes_per_sample(s->sample_fmt[1]);
-
-    if (s->sample_fmt[0] != AV_SAMPLE_FMT_S16) {
-        if (!(s->convert_ctx[0] = av_audio_convert_alloc(AV_SAMPLE_FMT_S16, 1,
-                                                         s->sample_fmt[0], 1, NULL, 0))) {
-            av_log(s, AV_LOG_ERROR,
-                   "Cannot convert %s sample format to s16 sample format\n",
-                   av_get_sample_fmt_name(s->sample_fmt[0]));
-            av_free(s);
-            return NULL;
-        }
-    }
-
-    if (s->sample_fmt[1] != AV_SAMPLE_FMT_S16) {
-        if (!(s->convert_ctx[1] = av_audio_convert_alloc(s->sample_fmt[1], 1,
-                                                         AV_SAMPLE_FMT_S16, 1, NULL, 0))) {
-            av_log(s, AV_LOG_ERROR,
-                   "Cannot convert s16 sample format to %s sample format\n",
-                   av_get_sample_fmt_name(s->sample_fmt[1]));
-            av_audio_convert_free(s->convert_ctx[0]);
-            av_free(s);
-            return NULL;
-        }
-    }
-
-    s->resample_context = av_resample_init(output_rate, input_rate,
-                                           filter_length, log2_phase_count,
-                                           linear, cutoff);
-
-    *(const AVClass**)s->resample_context = &audioresample_context_class;
-
-    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[MAX_CHANNELS];
-    short *bufout[MAX_CHANNELS];
-    short *buftmp2[MAX_CHANNELS], *buftmp3[MAX_CHANNELS];
-    short *output_bak = NULL;
-    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;
-    }
-
-    if (s->sample_fmt[0] != AV_SAMPLE_FMT_S16) {
-        int istride[1] = { s->sample_size[0] };
-        int ostride[1] = { 2 };
-        const void *ibuf[1] = { input };
-        void       *obuf[1];
-        unsigned input_size = nb_samples * s->input_channels * 2;
-
-        if (!s->buffer_size[0] || s->buffer_size[0] < input_size) {
-            av_free(s->buffer[0]);
-            s->buffer_size[0] = input_size;
-            s->buffer[0] = av_malloc(s->buffer_size[0]);
-            if (!s->buffer[0]) {
-                av_log(s->resample_context, AV_LOG_ERROR, "Could not allocate buffer\n");
-                return 0;
-            }
-        }
-
-        obuf[0] = s->buffer[0];
-
-        if (av_audio_convert(s->convert_ctx[0], obuf, ostride,
-                             ibuf, istride, nb_samples * s->input_channels) < 0) {
-            av_log(s->resample_context, AV_LOG_ERROR,
-                   "Audio sample format conversion failed\n");
-            return 0;
-        }
-
-        input = s->buffer[0];
-    }
-
-    lenout = 4 * nb_samples * s->ratio + 16;
-
-    if (s->sample_fmt[1] != AV_SAMPLE_FMT_S16) {
-        int out_size = lenout * av_get_bytes_per_sample(s->sample_fmt[1]) *
-                       s->output_channels;
-        output_bak = output;
-
-        if (!s->buffer_size[1] || s->buffer_size[1] < out_size) {
-            av_free(s->buffer[1]);
-            s->buffer_size[1] = out_size;
-            s->buffer[1] = av_malloc(s->buffer_size[1]);
-            if (!s->buffer[1]) {
-                av_log(s->resample_context, AV_LOG_ERROR, "Could not allocate buffer\n");
-                return 0;
-            }
-        }
-
-        output = s->buffer[1];
-    }
-
-    /* XXX: move those malloc to resample init code */
-    for (i = 0; i < s->filter_channels; i++) {
-        bufin[i] = 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;
-        bufout[i] = 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 >= s->input_channels && s->input_channels >= 2) {
-        for (i = 0; i < s->input_channels; i++) {
-            buftmp3[i] = bufout[i];
-        }
-        deinterleave(buftmp2, input, s->input_channels, 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 == 6 && s->input_channels == 2) {
-        ac3_5p1_mux(output, buftmp3[0], buftmp3[1], nb_samples1);
-    } else if (s->output_channels == s->input_channels && s->input_channels >= 2) {
-        interleave(output, buftmp3, s->output_channels, nb_samples1);
-    }
-
-    if (s->sample_fmt[1] != AV_SAMPLE_FMT_S16) {
-        int istride[1] = { 2 };
-        int ostride[1] = { s->sample_size[1] };
-        const void *ibuf[1] = { output };
-        void       *obuf[1] = { output_bak };
-
-        if (av_audio_convert(s->convert_ctx[1], obuf, ostride,
-                             ibuf, istride, nb_samples1 * s->output_channels) < 0) {
-            av_log(s->resample_context, AV_LOG_ERROR,
-                   "Audio sample format conversion failed\n");
-            return 0;
-        }
-    }
-
-    for (i = 0; i < s->filter_channels; i++) {
-        av_free(bufin[i]);
-        av_free(bufout[i]);
-    }
-
-    return nb_samples1;
-}
-
-void audio_resample_close(ReSampleContext *s)
-{
-    int i;
-    av_resample_close(s->resample_context);
-    for (i = 0; i < s->filter_channels; i++)
-        av_freep(&s->temp[i]);
-    av_freep(&s->buffer[0]);
-    av_freep(&s->buffer[1]);
-    av_audio_convert_free(s->convert_ctx[0]);
-    av_audio_convert_free(s->convert_ctx[1]);
-    av_free(s);
-}
-
-#endif