/*
  * Copyright (c) 2011 Stefano Sabatini
  * 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
  */
 
 /**
  * @file
  * audio volume filter
  */
 

 #include "libavutil/channel_layout.h"

 #include "libavutil/common.h"
 #include "libavutil/eval.h"
 #include "libavutil/float_dsp.h"

 #include "libavutil/intreadwrite.h"

 #include "libavutil/opt.h"

 #include "libavutil/replaygain.h"
 

 #include "audio.h"
 #include "avfilter.h"
 #include "formats.h"
 #include "internal.h"
 #include "af_volume.h"
 

 static const char * const precision_str[] = {

     "fixed", "float", "double"
 };
 

 static const char *const var_names[] = {
     "n",                   ///< frame number (starting at zero)
     "nb_channels",         ///< number of channels
     "nb_consumed_samples", ///< number of samples consumed by the filter
     "nb_samples",          ///< number of samples in the current frame
     "pos",                 ///< position in the file of the frame
     "pts",                 ///< frame presentation timestamp
     "sample_rate",         ///< sample rate
     "startpts",            ///< PTS at start of stream
     "startt",              ///< time at start of stream
     "t",                   ///< time in the file of the frame
     "tb",                  ///< timebase
     "volume",              ///< last set value
     NULL
 };
 

 #define OFFSET(x) offsetof(VolumeContext, x)
 #define A AV_OPT_FLAG_AUDIO_PARAM

 #define F AV_OPT_FLAG_FILTERING_PARAM

 static const AVOption volume_options[] = {

     { "volume", "set volume adjustment expression",
             OFFSET(volume_expr), AV_OPT_TYPE_STRING, { .str = "1.0" }, .flags = A|F },

     { "precision", "select mathematical precision",

             OFFSET(precision), AV_OPT_TYPE_INT, { .i64 = PRECISION_FLOAT }, PRECISION_FIXED, PRECISION_DOUBLE, A|F, "precision" },
         { "fixed",  "select 8-bit fixed-point",     0, AV_OPT_TYPE_CONST, { .i64 = PRECISION_FIXED  }, INT_MIN, INT_MAX, A|F, "precision" },
         { "float",  "select 32-bit floating-point", 0, AV_OPT_TYPE_CONST, { .i64 = PRECISION_FLOAT  }, INT_MIN, INT_MAX, A|F, "precision" },
         { "double", "select 64-bit floating-point", 0, AV_OPT_TYPE_CONST, { .i64 = PRECISION_DOUBLE }, INT_MIN, INT_MAX, A|F, "precision" },

     { "eval", "specify when to evaluate expressions", OFFSET(eval_mode), AV_OPT_TYPE_INT, {.i64 = EVAL_MODE_ONCE}, 0, EVAL_MODE_NB-1, .flags = A|F, "eval" },
          { "once",  "eval volume expression once", 0, AV_OPT_TYPE_CONST, {.i64=EVAL_MODE_ONCE},  .flags = A|F, .unit = "eval" },
          { "frame", "eval volume expression per-frame",                  0, AV_OPT_TYPE_CONST, {.i64=EVAL_MODE_FRAME}, .flags = A|F, .unit = "eval" },

     { "replaygain", "Apply replaygain side data when present",
             OFFSET(replaygain), AV_OPT_TYPE_INT, { .i64 = REPLAYGAIN_DROP }, REPLAYGAIN_DROP, REPLAYGAIN_ALBUM, A, "replaygain" },
         { "drop",   "replaygain side data is dropped", 0, AV_OPT_TYPE_CONST, { .i64 = REPLAYGAIN_DROP   }, 0, 0, A, "replaygain" },
         { "ignore", "replaygain side data is ignored", 0, AV_OPT_TYPE_CONST, { .i64 = REPLAYGAIN_IGNORE }, 0, 0, A, "replaygain" },
         { "track",  "track gain is preferred",         0, AV_OPT_TYPE_CONST, { .i64 = REPLAYGAIN_TRACK  }, 0, 0, A, "replaygain" },
         { "album",  "album gain is preferred",         0, AV_OPT_TYPE_CONST, { .i64 = REPLAYGAIN_ALBUM  }, 0, 0, A, "replaygain" },

     { "replaygain_preamp", "Apply replaygain pre-amplification",
             OFFSET(replaygain_preamp), AV_OPT_TYPE_DOUBLE, { .dbl = 0.0 }, -15.0, 15.0, A },

     { "replaygain_noclip", "Apply replaygain clipping prevention",
             OFFSET(replaygain_noclip), AV_OPT_TYPE_INT, { .i64 = 1 }, 0, 1, A },

     { NULL },

 };
 

 AVFILTER_DEFINE_CLASS(volume);

 static int set_expr(AVExpr **pexpr, const char *expr, void *log_ctx)

 {

     int ret;
     AVExpr *old = NULL;
 
     if (*pexpr)
         old = *pexpr;
     ret = av_expr_parse(pexpr, expr, var_names,
                         NULL, NULL, NULL, NULL, 0, log_ctx);
     if (ret < 0) {
         av_log(log_ctx, AV_LOG_ERROR,
                "Error when evaluating the volume expression '%s'\n", expr);
         *pexpr = old;
         return ret;

     }
 

     av_expr_free(old);

     return 0;

 }
 

 static av_cold int init(AVFilterContext *ctx)
 {
     VolumeContext *vol = ctx->priv;

 
     vol->fdsp = avpriv_float_dsp_alloc(0);
     if (!vol->fdsp)
         return AVERROR(ENOMEM);
 

     return set_expr(&vol->volume_pexpr, vol->volume_expr, ctx);
 }
 
 static av_cold void uninit(AVFilterContext *ctx)
 {
     VolumeContext *vol = ctx->priv;
     av_expr_free(vol->volume_pexpr);
     av_opt_free(vol);

     av_freep(&vol->fdsp);

 }
 

 static int query_formats(AVFilterContext *ctx)
 {
     VolumeContext *vol = ctx->priv;
     AVFilterFormats *formats = NULL;
     AVFilterChannelLayouts *layouts;
     static const enum AVSampleFormat sample_fmts[][7] = {

         [PRECISION_FIXED] = {

             AV_SAMPLE_FMT_U8,
             AV_SAMPLE_FMT_U8P,
             AV_SAMPLE_FMT_S16,
             AV_SAMPLE_FMT_S16P,
             AV_SAMPLE_FMT_S32,
             AV_SAMPLE_FMT_S32P,
             AV_SAMPLE_FMT_NONE
         },

         [PRECISION_FLOAT] = {

             AV_SAMPLE_FMT_FLT,
             AV_SAMPLE_FMT_FLTP,
             AV_SAMPLE_FMT_NONE
         },

         [PRECISION_DOUBLE] = {

             AV_SAMPLE_FMT_DBL,
             AV_SAMPLE_FMT_DBLP,
             AV_SAMPLE_FMT_NONE
         }
     };
 

     layouts = ff_all_channel_counts();

     if (!layouts)
         return AVERROR(ENOMEM);
     ff_set_common_channel_layouts(ctx, layouts);
 
     formats = ff_make_format_list(sample_fmts[vol->precision]);
     if (!formats)
         return AVERROR(ENOMEM);
     ff_set_common_formats(ctx, formats);
 
     formats = ff_all_samplerates();
     if (!formats)
         return AVERROR(ENOMEM);
     ff_set_common_samplerates(ctx, formats);
 
     return 0;
 }
 
 static inline void scale_samples_u8(uint8_t *dst, const uint8_t *src,
                                     int nb_samples, int volume)
 {
     int i;
     for (i = 0; i < nb_samples; i++)
         dst[i] = av_clip_uint8(((((int64_t)src[i] - 128) * volume + 128) >> 8) + 128);
 }
 
 static inline void scale_samples_u8_small(uint8_t *dst, const uint8_t *src,
                                           int nb_samples, int volume)
 {
     int i;
     for (i = 0; i < nb_samples; i++)
         dst[i] = av_clip_uint8((((src[i] - 128) * volume + 128) >> 8) + 128);
 }
 
 static inline void scale_samples_s16(uint8_t *dst, const uint8_t *src,
                                      int nb_samples, int volume)
 {
     int i;
     int16_t *smp_dst       = (int16_t *)dst;
     const int16_t *smp_src = (const int16_t *)src;
     for (i = 0; i < nb_samples; i++)
         smp_dst[i] = av_clip_int16(((int64_t)smp_src[i] * volume + 128) >> 8);
 }
 
 static inline void scale_samples_s16_small(uint8_t *dst, const uint8_t *src,
                                            int nb_samples, int volume)
 {
     int i;
     int16_t *smp_dst       = (int16_t *)dst;
     const int16_t *smp_src = (const int16_t *)src;
     for (i = 0; i < nb_samples; i++)
         smp_dst[i] = av_clip_int16((smp_src[i] * volume + 128) >> 8);
 }
 
 static inline void scale_samples_s32(uint8_t *dst, const uint8_t *src,
                                      int nb_samples, int volume)
 {
     int i;
     int32_t *smp_dst       = (int32_t *)dst;
     const int32_t *smp_src = (const int32_t *)src;
     for (i = 0; i < nb_samples; i++)
         smp_dst[i] = av_clipl_int32((((int64_t)smp_src[i] * volume + 128) >> 8));
 }
 

 static av_cold void volume_init(VolumeContext *vol)

 {
     vol->samples_align = 1;
 
     switch (av_get_packed_sample_fmt(vol->sample_fmt)) {
     case AV_SAMPLE_FMT_U8:
         if (vol->volume_i < 0x1000000)
             vol->scale_samples = scale_samples_u8_small;
         else
             vol->scale_samples = scale_samples_u8;
         break;
     case AV_SAMPLE_FMT_S16:
         if (vol->volume_i < 0x10000)
             vol->scale_samples = scale_samples_s16_small;
         else
             vol->scale_samples = scale_samples_s16;
         break;
     case AV_SAMPLE_FMT_S32:
         vol->scale_samples = scale_samples_s32;
         break;
     case AV_SAMPLE_FMT_FLT:
         vol->samples_align = 4;
         break;
     case AV_SAMPLE_FMT_DBL:
         vol->samples_align = 8;
         break;
     }

 
     if (ARCH_X86)
         ff_volume_init_x86(vol);

 }
 

 static int set_volume(AVFilterContext *ctx)
 {
     VolumeContext *vol = ctx->priv;
 
     vol->volume = av_expr_eval(vol->volume_pexpr, vol->var_values, NULL);
     if (isnan(vol->volume)) {
         if (vol->eval_mode == EVAL_MODE_ONCE) {
             av_log(ctx, AV_LOG_ERROR, "Invalid value NaN for volume\n");
             return AVERROR(EINVAL);
         } else {
             av_log(ctx, AV_LOG_WARNING, "Invalid value NaN for volume, setting to 0\n");
             vol->volume = 0;
         }
     }
     vol->var_values[VAR_VOLUME] = vol->volume;
 

     av_log(ctx, AV_LOG_VERBOSE, "n:%f t:%f pts:%f precision:%s ",
            vol->var_values[VAR_N], vol->var_values[VAR_T], vol->var_values[VAR_PTS],
            precision_str[vol->precision]);
 

     if (vol->precision == PRECISION_FIXED) {
         vol->volume_i = (int)(vol->volume * 256 + 0.5);
         vol->volume   = vol->volume_i / 256.0;

         av_log(ctx, AV_LOG_VERBOSE, "volume_i:%d/255 ", vol->volume_i);

     }

     av_log(ctx, AV_LOG_VERBOSE, "volume:%f volume_dB:%f\n",
            vol->volume, 20.0*log(vol->volume)/M_LN10);

 
     volume_init(vol);
     return 0;
 }
 

 static int config_output(AVFilterLink *outlink)
 {
     AVFilterContext *ctx = outlink->src;
     VolumeContext *vol   = ctx->priv;
     AVFilterLink *inlink = ctx->inputs[0];
 
     vol->sample_fmt = inlink->format;

     vol->channels   = inlink->channels;

     vol->planes     = av_sample_fmt_is_planar(inlink->format) ? vol->channels : 1;
 

     vol->var_values[VAR_N] =
     vol->var_values[VAR_NB_CONSUMED_SAMPLES] =
     vol->var_values[VAR_NB_SAMPLES] =
     vol->var_values[VAR_POS] =
     vol->var_values[VAR_PTS] =
     vol->var_values[VAR_STARTPTS] =
     vol->var_values[VAR_STARTT] =
     vol->var_values[VAR_T] =
     vol->var_values[VAR_VOLUME] = NAN;
 
     vol->var_values[VAR_NB_CHANNELS] = inlink->channels;
     vol->var_values[VAR_TB]          = av_q2d(inlink->time_base);
     vol->var_values[VAR_SAMPLE_RATE] = inlink->sample_rate;
 
     av_log(inlink->src, AV_LOG_VERBOSE, "tb:%f sample_rate:%f nb_channels:%f\n",
            vol->var_values[VAR_TB],
            vol->var_values[VAR_SAMPLE_RATE],
            vol->var_values[VAR_NB_CHANNELS]);
 
     return set_volume(ctx);

 }
 

 static int process_command(AVFilterContext *ctx, const char *cmd, const char *args,
                            char *res, int res_len, int flags)
 {
     VolumeContext *vol = ctx->priv;
     int ret = AVERROR(ENOSYS);
 
     if (!strcmp(cmd, "volume")) {
         if ((ret = set_expr(&vol->volume_pexpr, args, ctx)) < 0)
             return ret;
         if (vol->eval_mode == EVAL_MODE_ONCE)
             set_volume(ctx);
     }
 
     return ret;
 }
 

 #define D2TS(d)  (isnan(d) ? AV_NOPTS_VALUE : (int64_t)(d))
 #define TS2D(ts) ((ts) == AV_NOPTS_VALUE ? NAN : (double)(ts))
 #define TS2T(ts, tb) ((ts) == AV_NOPTS_VALUE ? NAN : (double)(ts)*av_q2d(tb))
 

 static int filter_frame(AVFilterLink *inlink, AVFrame *buf)

 {

     AVFilterContext *ctx = inlink->dst;

     VolumeContext *vol    = inlink->dst->priv;
     AVFilterLink *outlink = inlink->dst->outputs[0];

     int nb_samples        = buf->nb_samples;
     AVFrame *out_buf;

     int64_t pos;

     AVFrameSideData *sd = av_frame_get_side_data(buf, AV_FRAME_DATA_REPLAYGAIN);

     int ret;

     if (sd && vol->replaygain != REPLAYGAIN_IGNORE) {
         if (vol->replaygain != REPLAYGAIN_DROP) {
             AVReplayGain *replaygain = (AVReplayGain*)sd->data;

             int32_t gain  = 100000;
             uint32_t peak = 100000;
             float g, p;

 
             if (vol->replaygain == REPLAYGAIN_TRACK &&

                 replaygain->track_gain != INT32_MIN) {

                 gain = replaygain->track_gain;

 
                 if (replaygain->track_peak != 0)
                     peak = replaygain->track_peak;
             } else if (replaygain->album_gain != INT32_MIN) {

                 gain = replaygain->album_gain;

 
                 if (replaygain->album_peak != 0)
                     peak = replaygain->album_peak;
             } else {

                 av_log(inlink->dst, AV_LOG_WARNING, "Both ReplayGain gain "
                        "values are unknown.\n");
             }
             g = gain / 100000.0f;

             p = peak / 100000.0f;

 
             av_log(inlink->dst, AV_LOG_VERBOSE,
                    "Using gain %f dB from replaygain side data.\n", g);
 

             vol->volume   = pow(10, (g + vol->replaygain_preamp) / 20);

             if (vol->replaygain_noclip)
                 vol->volume = FFMIN(vol->volume, 1.0 / p);

             vol->volume_i = (int)(vol->volume * 256 + 0.5);
 
             volume_init(vol);
         }
         av_frame_remove_side_data(buf, AV_FRAME_DATA_REPLAYGAIN);
     }
 

     if (isnan(vol->var_values[VAR_STARTPTS])) {
         vol->var_values[VAR_STARTPTS] = TS2D(buf->pts);
         vol->var_values[VAR_STARTT  ] = TS2T(buf->pts, inlink->time_base);
     }
     vol->var_values[VAR_PTS] = TS2D(buf->pts);
     vol->var_values[VAR_T  ] = TS2T(buf->pts, inlink->time_base);
     vol->var_values[VAR_N  ] = inlink->frame_count;

     pos = av_frame_get_pkt_pos(buf);
     vol->var_values[VAR_POS] = pos == -1 ? NAN : pos;
     if (vol->eval_mode == EVAL_MODE_FRAME)
         set_volume(ctx);
 
     if (vol->volume == 1.0 || vol->volume_i == 256) {
         out_buf = buf;
         goto end;
     }

 
     /* do volume scaling in-place if input buffer is writable */

     if (av_frame_is_writable(buf)) {

         out_buf = buf;
     } else {

         out_buf = ff_get_audio_buffer(inlink, nb_samples);

         if (!out_buf)
             return AVERROR(ENOMEM);

         ret = av_frame_copy_props(out_buf, buf);
         if (ret < 0) {
             av_frame_free(&out_buf);
             av_frame_free(&buf);
             return ret;
         }

     }
 
     if (vol->precision != PRECISION_FIXED || vol->volume_i > 0) {
         int p, plane_samples;
 
         if (av_sample_fmt_is_planar(buf->format))
             plane_samples = FFALIGN(nb_samples, vol->samples_align);
         else
             plane_samples = FFALIGN(nb_samples * vol->channels, vol->samples_align);
 
         if (vol->precision == PRECISION_FIXED) {
             for (p = 0; p < vol->planes; p++) {
                 vol->scale_samples(out_buf->extended_data[p],
                                    buf->extended_data[p], plane_samples,
                                    vol->volume_i);
             }
         } else if (av_get_packed_sample_fmt(vol->sample_fmt) == AV_SAMPLE_FMT_FLT) {
             for (p = 0; p < vol->planes; p++) {

                 vol->fdsp->vector_fmul_scalar((float *)out_buf->extended_data[p],

                                              (const float *)buf->extended_data[p],
                                              vol->volume, plane_samples);
             }
         } else {
             for (p = 0; p < vol->planes; p++) {

                 vol->fdsp->vector_dmul_scalar((double *)out_buf->extended_data[p],

                                              (const double *)buf->extended_data[p],
                                              vol->volume, plane_samples);
             }
         }
     }
 

     emms_c();
 

     if (buf != out_buf)

         av_frame_free(&buf);

 end:

     vol->var_values[VAR_NB_CONSUMED_SAMPLES] += out_buf->nb_samples;

     return ff_filter_frame(outlink, out_buf);
 }
 
 static const AVFilterPad avfilter_af_volume_inputs[] = {
     {
         .name           = "default",
         .type           = AVMEDIA_TYPE_AUDIO,
         .filter_frame   = filter_frame,
     },
     { NULL }
 };
 
 static const AVFilterPad avfilter_af_volume_outputs[] = {
     {
         .name         = "default",
         .type         = AVMEDIA_TYPE_AUDIO,
         .config_props = config_output,
     },
     { NULL }
 };
 

 AVFilter ff_af_volume = {

     .name           = "volume",

     .description    = NULL_IF_CONFIG_SMALL("Change input volume."),
     .query_formats  = query_formats,
     .priv_size      = sizeof(VolumeContext),

     .priv_class     = &volume_class,

     .init           = init,

     .uninit         = uninit,

     .inputs         = avfilter_af_volume_inputs,
     .outputs        = avfilter_af_volume_outputs,

     .flags          = AVFILTER_FLAG_SUPPORT_TIMELINE_GENERIC,

     .process_command = process_command,

 };