/*
  * Copyright (c) 2004 Michael Niedermayer <michaelni@gmx.at>
  * Copyright (c) 2012 Justin Ruggles <justin.ruggles@gmail.com>
  *

  * 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
  */
 

 #include "libavutil/common.h"

 #include "libavutil/libm.h"
 #include "libavutil/log.h"
 #include "internal.h"

 #include "resample.h"

 #include "audio_data.h"
 

 
 /* double template */
 #define CONFIG_RESAMPLE_DBL
 #include "resample_template.c"
 #undef CONFIG_RESAMPLE_DBL
 
 /* float template */
 #define CONFIG_RESAMPLE_FLT
 #include "resample_template.c"
 #undef CONFIG_RESAMPLE_FLT
 
 /* s32 template */
 #define CONFIG_RESAMPLE_S32
 #include "resample_template.c"
 #undef CONFIG_RESAMPLE_S32
 
 /* s16 template */
 #include "resample_template.c"
 
 

 /* 0th order modified bessel function of the first kind. */

 static double bessel(double x)
 {
     double v     = 1;
     double lastv = 0;
     double t     = 1;
     int i;
 
     x = x * x / 4;
     for (i = 1; v != lastv; i++) {
         lastv = v;
         t    *= x / (i * i);
         v    += t;
     }
     return v;
 }
 

 /* Build a polyphase filterbank. */

 static int build_filter(ResampleContext *c, double factor)

 {
     int ph, i;

     double x, y, w;

     double *tab;

     int tap_count    = c->filter_length;
     int phase_count  = 1 << c->phase_shift;

     const int center = (tap_count - 1) / 2;
 
     tab = av_malloc(tap_count * sizeof(*tab));
     if (!tab)
         return AVERROR(ENOMEM);
 
     for (ph = 0; ph < phase_count; ph++) {
         double norm = 0;
         for (i = 0; i < tap_count; i++) {
             x = M_PI * ((double)(i - center) - (double)ph / phase_count) * factor;
             if (x == 0) y = 1.0;
             else        y = sin(x) / x;

             switch (c->filter_type) {

             case AV_RESAMPLE_FILTER_TYPE_CUBIC: {

                 const float d = -0.5; //first order derivative = -0.5
                 x = fabs(((double)(i - center) - (double)ph / phase_count) * factor);
                 if (x < 1.0) y = 1 - 3 * x*x + 2 * x*x*x + d * (                -x*x + x*x*x);
                 else         y =                           d * (-4 + 8 * x - 5 * x*x + x*x*x);
                 break;
             }

             case AV_RESAMPLE_FILTER_TYPE_BLACKMAN_NUTTALL:

                 w  = 2.0 * x / (factor * tap_count) + M_PI;
                 y *= 0.3635819 - 0.4891775 * cos(    w) +
                                  0.1365995 * cos(2 * w) -
                                  0.0106411 * cos(3 * w);
                 break;

             case AV_RESAMPLE_FILTER_TYPE_KAISER:

                 w  = 2.0 * x / (factor * tap_count * M_PI);

                 y *= bessel(c->kaiser_beta * sqrt(FFMAX(1 - w * w, 0)));

                 break;
             }
 
             tab[i] = y;
             norm  += y;
         }
         /* normalize so that an uniform color remains the same */

         for (i = 0; i < tap_count; i++)
             tab[i] = tab[i] / norm;
 
         c->set_filter(c->filter_bank, tab, ph, tap_count);

     }
 
     av_free(tab);
     return 0;
 }
 
 ResampleContext *ff_audio_resample_init(AVAudioResampleContext *avr)
 {
     ResampleContext *c;
     int out_rate    = avr->out_sample_rate;
     int in_rate     = avr->in_sample_rate;
     double factor   = FFMIN(out_rate * avr->cutoff / in_rate, 1.0);
     int phase_count = 1 << avr->phase_shift;

     int felem_size;

     if (avr->internal_sample_fmt != AV_SAMPLE_FMT_S16P &&
         avr->internal_sample_fmt != AV_SAMPLE_FMT_S32P &&
         avr->internal_sample_fmt != AV_SAMPLE_FMT_FLTP &&
         avr->internal_sample_fmt != AV_SAMPLE_FMT_DBLP) {

         av_log(avr, AV_LOG_ERROR, "Unsupported internal format for "
                "resampling: %s\n",
                av_get_sample_fmt_name(avr->internal_sample_fmt));
         return NULL;
     }
     c = av_mallocz(sizeof(*c));
     if (!c)
         return NULL;
 
     c->avr           = avr;
     c->phase_shift   = avr->phase_shift;
     c->phase_mask    = phase_count - 1;
     c->linear        = avr->linear_interp;
     c->filter_length = FFMAX((int)ceil(avr->filter_size / factor), 1);

     c->filter_type   = avr->filter_type;
     c->kaiser_beta   = avr->kaiser_beta;

     switch (avr->internal_sample_fmt) {
     case AV_SAMPLE_FMT_DBLP:

         c->resample_one  = c->linear ? resample_linear_dbl : resample_one_dbl;

         c->resample_nearest = resample_nearest_dbl;

         c->set_filter    = set_filter_dbl;
         break;
     case AV_SAMPLE_FMT_FLTP:

         c->resample_one  = c->linear ? resample_linear_flt : resample_one_flt;

         c->resample_nearest = resample_nearest_flt;

         c->set_filter    = set_filter_flt;
         break;
     case AV_SAMPLE_FMT_S32P:

         c->resample_one  = c->linear ? resample_linear_s32 : resample_one_s32;

         c->resample_nearest = resample_nearest_s32;

         c->set_filter    = set_filter_s32;
         break;
     case AV_SAMPLE_FMT_S16P:

         c->resample_one  = c->linear ? resample_linear_s16 : resample_one_s16;

         c->resample_nearest = resample_nearest_s16;

         c->set_filter    = set_filter_s16;
         break;
     }
 

     if (ARCH_AARCH64)
         ff_audio_resample_init_aarch64(c, avr->internal_sample_fmt);

     if (ARCH_ARM)
         ff_audio_resample_init_arm(c, avr->internal_sample_fmt);

     felem_size = av_get_bytes_per_sample(avr->internal_sample_fmt);
     c->filter_bank = av_mallocz(c->filter_length * (phase_count + 1) * felem_size);

     if (!c->filter_bank)
         goto error;
 

     if (build_filter(c, factor) < 0)

         goto error;
 

     memcpy(&c->filter_bank[(c->filter_length * phase_count + 1) * felem_size],
            c->filter_bank, (c->filter_length - 1) * felem_size);
     memcpy(&c->filter_bank[c->filter_length * phase_count * felem_size],
            &c->filter_bank[(c->filter_length - 1) * felem_size], felem_size);

 
     c->compensation_distance = 0;
     if (!av_reduce(&c->src_incr, &c->dst_incr, out_rate,
                    in_rate * (int64_t)phase_count, INT32_MAX / 2))
         goto error;
     c->ideal_dst_incr = c->dst_incr;
 

     c->padding_size   = (c->filter_length - 1) / 2;

     c->initial_padding_filled = 0;
     c->index = 0;

     c->frac  = 0;
 
     /* allocate internal buffer */

     c->buffer = ff_audio_data_alloc(avr->resample_channels, c->padding_size,

                                     avr->internal_sample_fmt,
                                     "resample buffer");
     if (!c->buffer)
         goto error;

     c->buffer->nb_samples      = c->padding_size;
     c->initial_padding_samples = c->padding_size;

 
     av_log(avr, AV_LOG_DEBUG, "resample: %s from %d Hz to %d Hz\n",
            av_get_sample_fmt_name(avr->internal_sample_fmt),
            avr->in_sample_rate, avr->out_sample_rate);
 
     return c;
 
 error:
     ff_audio_data_free(&c->buffer);
     av_free(c->filter_bank);
     av_free(c);
     return NULL;
 }
 
 void ff_audio_resample_free(ResampleContext **c)
 {
     if (!*c)
         return;
     ff_audio_data_free(&(*c)->buffer);
     av_free((*c)->filter_bank);
     av_freep(c);
 }
 
 int avresample_set_compensation(AVAudioResampleContext *avr, int sample_delta,
                                 int compensation_distance)
 {
     ResampleContext *c;
     AudioData *fifo_buf = NULL;
     int ret = 0;
 
     if (compensation_distance < 0)
         return AVERROR(EINVAL);
     if (!compensation_distance && sample_delta)
         return AVERROR(EINVAL);
 
     if (!avr->resample_needed) {

 #if FF_API_RESAMPLE_CLOSE_OPEN
         /* if resampling was not enabled previously, re-initialize the
            AVAudioResampleContext and force resampling */

         int fifo_samples;

         int restore_matrix = 0;

         double matrix[AVRESAMPLE_MAX_CHANNELS * AVRESAMPLE_MAX_CHANNELS] = { 0 };
 
         /* buffer any remaining samples in the output FIFO before closing */
         fifo_samples = av_audio_fifo_size(avr->out_fifo);
         if (fifo_samples > 0) {
             fifo_buf = ff_audio_data_alloc(avr->out_channels, fifo_samples,
                                            avr->out_sample_fmt, NULL);
             if (!fifo_buf)
                 return AVERROR(EINVAL);
             ret = ff_audio_data_read_from_fifo(avr->out_fifo, fifo_buf,
                                                fifo_samples);
             if (ret < 0)
                 goto reinit_fail;
         }
         /* save the channel mixing matrix */

         if (avr->am) {
             ret = avresample_get_matrix(avr, matrix, AVRESAMPLE_MAX_CHANNELS);
             if (ret < 0)
                 goto reinit_fail;
             restore_matrix = 1;
         }

 
         /* close the AVAudioResampleContext */
         avresample_close(avr);
 
         avr->force_resampling = 1;
 
         /* restore the channel mixing matrix */

         if (restore_matrix) {
             ret = avresample_set_matrix(avr, matrix, AVRESAMPLE_MAX_CHANNELS);
             if (ret < 0)
                 goto reinit_fail;
         }

 
         /* re-open the AVAudioResampleContext */
         ret = avresample_open(avr);
         if (ret < 0)
             goto reinit_fail;
 
         /* restore buffered samples to the output FIFO */
         if (fifo_samples > 0) {
             ret = ff_audio_data_add_to_fifo(avr->out_fifo, fifo_buf, 0,
                                             fifo_samples);
             if (ret < 0)
                 goto reinit_fail;
             ff_audio_data_free(&fifo_buf);
         }

 #else
         av_log(avr, AV_LOG_ERROR, "Unable to set resampling compensation\n");
         return AVERROR(EINVAL);
 #endif

     }
     c = avr->resample;
     c->compensation_distance = compensation_distance;
     if (compensation_distance) {
         c->dst_incr = c->ideal_dst_incr - c->ideal_dst_incr *
                       (int64_t)sample_delta / compensation_distance;
     } else {
         c->dst_incr = c->ideal_dst_incr;
     }
     return 0;
 
 reinit_fail:
     ff_audio_data_free(&fifo_buf);
     return ret;
 }
 

 static int resample(ResampleContext *c, void *dst, const void *src,

                     int *consumed, int src_size, int dst_size, int update_ctx,
                     int nearest_neighbour)

 {

     int dst_index;

     unsigned int index = c->index;

     int frac          = c->frac;
     int dst_incr_frac = c->dst_incr % c->src_incr;
     int dst_incr      = c->dst_incr / c->src_incr;
     int compensation_distance = c->compensation_distance;
 
     if (!dst != !src)
         return AVERROR(EINVAL);
 

     if (nearest_neighbour) {

         uint64_t index2 = ((uint64_t)index) << 32;

         int64_t incr   = (1LL << 32) * c->dst_incr / c->src_incr;
         dst_size       = FFMIN(dst_size,
                                (src_size-1-index) * (int64_t)c->src_incr /
                                c->dst_incr);
 
         if (dst) {
             for(dst_index = 0; dst_index < dst_size; dst_index++) {

                 c->resample_nearest(dst, dst_index, src, index2 >> 32);

                 index2 += incr;
             }
         } else {
             dst_index = dst_size;
         }
         index += dst_index * dst_incr;
         index += (frac + dst_index * (int64_t)dst_incr_frac) / c->src_incr;
         frac   = (frac + dst_index * (int64_t)dst_incr_frac) % c->src_incr;
     } else {
         for (dst_index = 0; dst_index < dst_size; dst_index++) {
             int sample_index = index >> c->phase_shift;
 

             if (sample_index + c->filter_length > src_size)

                 break;
 

             if (dst)

                 c->resample_one(c, dst, dst_index, src, index, frac);

 
             frac  += dst_incr_frac;
             index += dst_incr;
             if (frac >= c->src_incr) {
                 frac -= c->src_incr;
                 index++;
             }
             if (dst_index + 1 == compensation_distance) {
                 compensation_distance = 0;
                 dst_incr_frac = c->ideal_dst_incr % c->src_incr;
                 dst_incr      = c->ideal_dst_incr / c->src_incr;
             }
         }
     }
     if (consumed)

         *consumed = index >> c->phase_shift;

 
     if (update_ctx) {

         index &= c->phase_mask;

 
         if (compensation_distance) {
             compensation_distance -= dst_index;
             if (compensation_distance <= 0)
                 return AVERROR_BUG;
         }
         c->frac     = frac;
         c->index    = index;
         c->dst_incr = dst_incr_frac + c->src_incr*dst_incr;
         c->compensation_distance = compensation_distance;
     }
 
     return dst_index;
 }
 

 int ff_audio_resample(ResampleContext *c, AudioData *dst, AudioData *src)

 {

     int ch, in_samples, in_leftover, consumed = 0, out_samples = 0;

     int ret = AVERROR(EINVAL);

     int nearest_neighbour = (c->compensation_distance == 0 &&
                              c->filter_length == 1 &&
                              c->phase_shift == 0);

 
     in_samples  = src ? src->nb_samples : 0;
     in_leftover = c->buffer->nb_samples;
 
     /* add input samples to the internal buffer */
     if (src) {
         ret = ff_audio_data_combine(c->buffer, in_leftover, src, 0, in_samples);
         if (ret < 0)
             return ret;

     } else if (in_leftover <= c->final_padding_samples) {

         /* no remaining samples to flush */
         return 0;
     }
 

     if (!c->initial_padding_filled) {
         int bps = av_get_bytes_per_sample(c->avr->internal_sample_fmt);
         int i;
 

         if (src && c->buffer->nb_samples < 2 * c->padding_size)

             return 0;
 
         for (i = 0; i < c->padding_size; i++)

             for (ch = 0; ch < c->buffer->channels; ch++) {
                 if (c->buffer->nb_samples > 2 * c->padding_size - i) {
                     memcpy(c->buffer->data[ch] + bps * i,
                            c->buffer->data[ch] + bps * (2 * c->padding_size - i), bps);
                 } else {
                     memset(c->buffer->data[ch] + bps * i, 0, bps);
                 }
             }

         c->initial_padding_filled = 1;
     }
 

     if (!src && !c->final_padding_filled) {
         int bps = av_get_bytes_per_sample(c->avr->internal_sample_fmt);
         int i;
 

         ret = ff_audio_data_realloc(c->buffer,
                                     FFMAX(in_samples, in_leftover) +
                                     c->padding_size);

         if (ret < 0) {
             av_log(c->avr, AV_LOG_ERROR, "Error reallocating resampling buffer\n");
             return AVERROR(ENOMEM);
         }
 
         for (i = 0; i < c->padding_size; i++)
             for (ch = 0; ch < c->buffer->channels; ch++) {
                 if (in_leftover > i) {
                     memcpy(c->buffer->data[ch] + bps * (in_leftover + i),
                            c->buffer->data[ch] + bps * (in_leftover - i - 1),
                            bps);
                 } else {
                     memset(c->buffer->data[ch] + bps * (in_leftover + i),
                            0, bps);
                 }
             }
         c->buffer->nb_samples   += c->padding_size;
         c->final_padding_samples = c->padding_size;
         c->final_padding_filled  = 1;
     }
 
 

     /* calculate output size and reallocate output buffer if needed */
     /* TODO: try to calculate this without the dummy resample() run */
     if (!dst->read_only && dst->allow_realloc) {
         out_samples = resample(c, NULL, NULL, NULL, c->buffer->nb_samples,

                                INT_MAX, 0, nearest_neighbour);

         ret = ff_audio_data_realloc(dst, out_samples);
         if (ret < 0) {
             av_log(c->avr, AV_LOG_ERROR, "error reallocating output\n");
             return ret;
         }
     }
 
     /* resample each channel plane */
     for (ch = 0; ch < c->buffer->channels; ch++) {

         out_samples = resample(c, (void *)dst->data[ch],

                                (const void *)c->buffer->data[ch], &consumed,

                                c->buffer->nb_samples, dst->allocated_samples,

                                ch + 1 == c->buffer->channels, nearest_neighbour);

     }
     if (out_samples < 0) {
         av_log(c->avr, AV_LOG_ERROR, "error during resampling\n");
         return out_samples;
     }
 
     /* drain consumed samples from the internal buffer */

     ff_audio_data_drain(c->buffer, consumed);

     c->initial_padding_samples = FFMAX(c->initial_padding_samples - consumed, 0);

     av_log(c->avr, AV_LOG_TRACE, "resampled %d in + %d leftover to %d out + %d leftover\n",

             in_samples, in_leftover, out_samples, c->buffer->nb_samples);
 
     dst->nb_samples = out_samples;
     return 0;
 }
 
 int avresample_get_delay(AVAudioResampleContext *avr)
 {

     ResampleContext *c = avr->resample;
 

     if (!avr->resample_needed || !avr->resample)
         return 0;
 

     return FFMAX(c->buffer->nb_samples - c->padding_size, 0);

 }