/*
  * Copyright (c) 2012 Justin Ruggles <justin.ruggles@gmail.com>
  *
  * Triangular with Noise Shaping is based on opusfile.
  * Copyright (c) 1994-2012 by the Xiph.Org Foundation and contributors
  *
  * 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
  * Dithered Audio Sample Quantization
  *
  * Converts from dbl, flt, or s32 to s16 using dithering.
  */
 
 #include <math.h>
 #include <stdint.h>
 
 #include "libavutil/common.h"
 #include "libavutil/lfg.h"
 #include "libavutil/mem.h"
 #include "libavutil/samplefmt.h"
 #include "audio_convert.h"
 #include "dither.h"
 #include "internal.h"
 
 typedef struct DitherState {
     int mute;
     unsigned int seed;
     AVLFG lfg;
     float *noise_buf;
     int noise_buf_size;
     int noise_buf_ptr;
     float dither_a[4];
     float dither_b[4];
 } DitherState;
 
 struct DitherContext {
     DitherDSPContext  ddsp;
     enum AVResampleDitherMethod method;

     int apply_map;
     ChannelMapInfo *ch_map_info;

 
     int mute_dither_threshold;  // threshold for disabling dither
     int mute_reset_threshold;   // threshold for resetting noise shaping
     const float *ns_coef_b;     // noise shaping coeffs
     const float *ns_coef_a;     // noise shaping coeffs
 
     int channels;
     DitherState *state;         // dither states for each channel
 
     AudioData *flt_data;        // input data in fltp
     AudioData *s16_data;        // dithered output in s16p
     AudioConvert *ac_in;        // converter for input to fltp
     AudioConvert *ac_out;       // converter for s16p to s16 (if needed)
 
     void (*quantize)(int16_t *dst, const float *src, float *dither, int len);
     int samples_align;
 };
 
 /* mute threshold, in seconds */
 #define MUTE_THRESHOLD_SEC 0.000333
 
 /* scale factor for 16-bit output.
    The signal is attenuated slightly to avoid clipping */
 #define S16_SCALE 32753.0f
 
 /* scale to convert lfg from INT_MIN/INT_MAX to -0.5/0.5 */
 #define LFG_SCALE (1.0f / (2.0f * INT32_MAX))
 
 /* noise shaping coefficients */
 
 static const float ns_48_coef_b[4] = {
     2.2374f, -0.7339f, -0.1251f, -0.6033f
 };
 
 static const float ns_48_coef_a[4] = {
     0.9030f, 0.0116f, -0.5853f, -0.2571f
 };
 
 static const float ns_44_coef_b[4] = {
     2.2061f, -0.4707f, -0.2534f, -0.6213f
 };
 
 static const float ns_44_coef_a[4] = {
     1.0587f, 0.0676f, -0.6054f, -0.2738f
 };
 
 static void dither_int_to_float_rectangular_c(float *dst, int *src, int len)
 {
     int i;
     for (i = 0; i < len; i++)
         dst[i] = src[i] * LFG_SCALE;
 }
 
 static void dither_int_to_float_triangular_c(float *dst, int *src0, int len)
 {
     int i;
     int *src1  = src0 + len;
 
     for (i = 0; i < len; i++) {
         float r = src0[i] * LFG_SCALE;
         r      += src1[i] * LFG_SCALE;
         dst[i]  = r;
     }
 }
 
 static void quantize_c(int16_t *dst, const float *src, float *dither, int len)
 {
     int i;
     for (i = 0; i < len; i++)
         dst[i] = av_clip_int16(lrintf(src[i] * S16_SCALE + dither[i]));
 }
 
 #define SQRT_1_6 0.40824829046386301723f
 
 static void dither_highpass_filter(float *src, int len)
 {
     int i;
 
     /* filter is from libswresample in FFmpeg */
     for (i = 0; i < len - 2; i++)
         src[i] = (-src[i] + 2 * src[i + 1] - src[i + 2]) * SQRT_1_6;
 }
 
 static int generate_dither_noise(DitherContext *c, DitherState *state,
                                  int min_samples)
 {
     int i;
     int nb_samples  = FFALIGN(min_samples, 16) + 16;
     int buf_samples = nb_samples *
                       (c->method == AV_RESAMPLE_DITHER_RECTANGULAR ? 1 : 2);
     unsigned int *noise_buf_ui;
 
     av_freep(&state->noise_buf);
     state->noise_buf_size = state->noise_buf_ptr = 0;
 
     state->noise_buf = av_malloc(buf_samples * sizeof(*state->noise_buf));
     if (!state->noise_buf)
         return AVERROR(ENOMEM);
     state->noise_buf_size = FFALIGN(min_samples, 16);
     noise_buf_ui          = (unsigned int *)state->noise_buf;
 
     av_lfg_init(&state->lfg, state->seed);
     for (i = 0; i < buf_samples; i++)
         noise_buf_ui[i] = av_lfg_get(&state->lfg);
 
     c->ddsp.dither_int_to_float(state->noise_buf, noise_buf_ui, nb_samples);
 
     if (c->method == AV_RESAMPLE_DITHER_TRIANGULAR_HP)
         dither_highpass_filter(state->noise_buf, nb_samples);
 
     return 0;
 }
 
 static void quantize_triangular_ns(DitherContext *c, DitherState *state,
                                    int16_t *dst, const float *src,
                                    int nb_samples)
 {
     int i, j;
     float *dither = &state->noise_buf[state->noise_buf_ptr];
 
     if (state->mute > c->mute_reset_threshold)
         memset(state->dither_a, 0, sizeof(state->dither_a));
 
     for (i = 0; i < nb_samples; i++) {
         float err = 0;
         float sample = src[i] * S16_SCALE;
 
         for (j = 0; j < 4; j++) {
             err += c->ns_coef_b[j] * state->dither_b[j] -
                    c->ns_coef_a[j] * state->dither_a[j];
         }
         for (j = 3; j > 0; j--) {
             state->dither_a[j] = state->dither_a[j - 1];
             state->dither_b[j] = state->dither_b[j - 1];
         }
         state->dither_a[0] = err;
         sample -= err;
 
         if (state->mute > c->mute_dither_threshold) {
             dst[i]             = av_clip_int16(lrintf(sample));
             state->dither_b[0] = 0;
         } else {
             dst[i]             = av_clip_int16(lrintf(sample + dither[i]));
             state->dither_b[0] = av_clipf(dst[i] - sample, -1.5f, 1.5f);
         }
 
         state->mute++;
         if (src[i])
             state->mute = 0;
     }
 }
 
 static int convert_samples(DitherContext *c, int16_t **dst, float * const *src,
                            int channels, int nb_samples)
 {
     int ch, ret;
     int aligned_samples = FFALIGN(nb_samples, 16);
 
     for (ch = 0; ch < channels; ch++) {
         DitherState *state = &c->state[ch];
 
         if (state->noise_buf_size < aligned_samples) {
             ret = generate_dither_noise(c, state, nb_samples);
             if (ret < 0)
                 return ret;
         } else if (state->noise_buf_size - state->noise_buf_ptr < aligned_samples) {
             state->noise_buf_ptr = 0;
         }
 
         if (c->method == AV_RESAMPLE_DITHER_TRIANGULAR_NS) {
             quantize_triangular_ns(c, state, dst[ch], src[ch], nb_samples);
         } else {
             c->quantize(dst[ch], src[ch],
                         &state->noise_buf[state->noise_buf_ptr],
                         FFALIGN(nb_samples, c->samples_align));
         }
 
         state->noise_buf_ptr += aligned_samples;
     }
 
     return 0;
 }
 
 int ff_convert_dither(DitherContext *c, AudioData *dst, AudioData *src)
 {
     int ret;
     AudioData *flt_data;
 
     /* output directly to dst if it is planar */
     if (dst->sample_fmt == AV_SAMPLE_FMT_S16P)
         c->s16_data = dst;
     else {
         /* make sure s16_data is large enough for the output */
         ret = ff_audio_data_realloc(c->s16_data, src->nb_samples);
         if (ret < 0)
             return ret;
     }
 

     if (src->sample_fmt != AV_SAMPLE_FMT_FLTP || c->apply_map) {

         /* make sure flt_data is large enough for the input */
         ret = ff_audio_data_realloc(c->flt_data, src->nb_samples);
         if (ret < 0)
             return ret;
         flt_data = c->flt_data;

     }

     if (src->sample_fmt != AV_SAMPLE_FMT_FLTP) {

         /* convert input samples to fltp and scale to s16 range */
         ret = ff_audio_convert(c->ac_in, flt_data, src);
         if (ret < 0)
             return ret;

     } else if (c->apply_map) {
         ret = ff_audio_data_copy(flt_data, src, c->ch_map_info);
         if (ret < 0)
             return ret;

     } else {
         flt_data = src;
     }
 
     /* check alignment and padding constraints */
     if (c->method != AV_RESAMPLE_DITHER_TRIANGULAR_NS) {
         int ptr_align     = FFMIN(flt_data->ptr_align,     c->s16_data->ptr_align);
         int samples_align = FFMIN(flt_data->samples_align, c->s16_data->samples_align);
         int aligned_len   = FFALIGN(src->nb_samples, c->ddsp.samples_align);
 
         if (!(ptr_align % c->ddsp.ptr_align) && samples_align >= aligned_len) {
             c->quantize      = c->ddsp.quantize;
             c->samples_align = c->ddsp.samples_align;
         } else {
             c->quantize      = quantize_c;
             c->samples_align = 1;
         }
     }
 
     ret = convert_samples(c, (int16_t **)c->s16_data->data,
                           (float * const *)flt_data->data, src->channels,
                           src->nb_samples);
     if (ret < 0)
         return ret;
 
     c->s16_data->nb_samples = src->nb_samples;
 
     /* interleave output to dst if needed */
     if (dst->sample_fmt == AV_SAMPLE_FMT_S16) {
         ret = ff_audio_convert(c->ac_out, dst, c->s16_data);
         if (ret < 0)
             return ret;
     } else
         c->s16_data = NULL;
 
     return 0;
 }
 
 void ff_dither_free(DitherContext **cp)
 {
     DitherContext *c = *cp;
     int ch;
 
     if (!c)
         return;
     ff_audio_data_free(&c->flt_data);
     ff_audio_data_free(&c->s16_data);
     ff_audio_convert_free(&c->ac_in);
     ff_audio_convert_free(&c->ac_out);
     for (ch = 0; ch < c->channels; ch++)
         av_free(c->state[ch].noise_buf);
     av_free(c->state);
     av_freep(cp);
 }
 
 static void dither_init(DitherDSPContext *ddsp,
                         enum AVResampleDitherMethod method)
 {
     ddsp->quantize      = quantize_c;
     ddsp->ptr_align     = 1;
     ddsp->samples_align = 1;
 
     if (method == AV_RESAMPLE_DITHER_RECTANGULAR)
         ddsp->dither_int_to_float = dither_int_to_float_rectangular_c;
     else
         ddsp->dither_int_to_float = dither_int_to_float_triangular_c;

 
     if (ARCH_X86)
         ff_dither_init_x86(ddsp, method);

 }
 
 DitherContext *ff_dither_alloc(AVAudioResampleContext *avr,
                                enum AVSampleFormat out_fmt,
                                enum AVSampleFormat in_fmt,

                                int channels, int sample_rate, int apply_map)

 {
     AVLFG seed_gen;
     DitherContext *c;
     int ch;
 
     if (av_get_packed_sample_fmt(out_fmt) != AV_SAMPLE_FMT_S16 ||
         av_get_bytes_per_sample(in_fmt) <= 2) {
         av_log(avr, AV_LOG_ERROR, "dithering %s to %s is not supported\n",
                av_get_sample_fmt_name(in_fmt), av_get_sample_fmt_name(out_fmt));
         return NULL;
     }
 
     c = av_mallocz(sizeof(*c));
     if (!c)
         return NULL;
 

     c->apply_map = apply_map;
     if (apply_map)
         c->ch_map_info = &avr->ch_map_info;
 

     if (avr->dither_method == AV_RESAMPLE_DITHER_TRIANGULAR_NS &&
         sample_rate != 48000 && sample_rate != 44100) {
         av_log(avr, AV_LOG_WARNING, "sample rate must be 48000 or 44100 Hz "
                "for triangular_ns dither. using triangular_hp instead.\n");
         avr->dither_method = AV_RESAMPLE_DITHER_TRIANGULAR_HP;
     }
     c->method = avr->dither_method;
     dither_init(&c->ddsp, c->method);
 
     if (c->method == AV_RESAMPLE_DITHER_TRIANGULAR_NS) {
         if (sample_rate == 48000) {
             c->ns_coef_b = ns_48_coef_b;
             c->ns_coef_a = ns_48_coef_a;
         } else {
             c->ns_coef_b = ns_44_coef_b;
             c->ns_coef_a = ns_44_coef_a;
         }
     }
 
     /* Either s16 or s16p output format is allowed, but s16p is used
        internally, so we need to use a temp buffer and interleave if the output
        format is s16 */
     if (out_fmt != AV_SAMPLE_FMT_S16P) {
         c->s16_data = ff_audio_data_alloc(channels, 1024, AV_SAMPLE_FMT_S16P,
                                           "dither s16 buffer");
         if (!c->s16_data)
             goto fail;
 
         c->ac_out = ff_audio_convert_alloc(avr, out_fmt, AV_SAMPLE_FMT_S16P,

                                            channels, sample_rate, 0);

         if (!c->ac_out)
             goto fail;
     }
 

     if (in_fmt != AV_SAMPLE_FMT_FLTP || c->apply_map) {

         c->flt_data = ff_audio_data_alloc(channels, 1024, AV_SAMPLE_FMT_FLTP,
                                           "dither flt buffer");
         if (!c->flt_data)
             goto fail;

     }
     if (in_fmt != AV_SAMPLE_FMT_FLTP) {

         c->ac_in = ff_audio_convert_alloc(avr, AV_SAMPLE_FMT_FLTP, in_fmt,

                                           channels, sample_rate, c->apply_map);

         if (!c->ac_in)
             goto fail;
     }
 
     c->state = av_mallocz(channels * sizeof(*c->state));
     if (!c->state)
         goto fail;
     c->channels = channels;
 
     /* calculate thresholds for turning off dithering during periods of
        silence to avoid replacing digital silence with quiet dither noise */
     c->mute_dither_threshold = lrintf(sample_rate * MUTE_THRESHOLD_SEC);
     c->mute_reset_threshold  = c->mute_dither_threshold * 4;
 
     /* initialize dither states */
     av_lfg_init(&seed_gen, 0xC0FFEE);
     for (ch = 0; ch < channels; ch++) {
         DitherState *state = &c->state[ch];
         state->mute = c->mute_reset_threshold + 1;
         state->seed = av_lfg_get(&seed_gen);
         generate_dither_noise(c, state, FFMAX(32768, sample_rate / 2));
     }
 
     return c;
 
 fail:
     ff_dither_free(&c);
     return NULL;
 }