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  1. /*
  2.  * Copyright (c) 2011 Stefano Sabatini
  3.  * Copyright (c) 2012 Justin Ruggles <justin.ruggles@gmail.com>
  4.  *
  5.  * This file is part of FFmpeg.
  6.  *
  7.  * FFmpeg is free software; you can redistribute it and/or
  8.  * modify it under the terms of the GNU Lesser General Public
  9.  * License as published by the Free Software Foundation; either
  10.  * version 2.1 of the License, or (at your option) any later version.
  11.  *
  12.  * FFmpeg is distributed in the hope that it will be useful,
  13.  * but WITHOUT ANY WARRANTY; without even the implied warranty of
  14.  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
  15.  * Lesser General Public License for more details.
  16.  *
  17.  * You should have received a copy of the GNU Lesser General Public
  18.  * License along with FFmpeg; if not, write to the Free Software
  19.  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  20.  */
  21.  
  22. /**
  23.  * @file
  24.  * audio volume filter
  25.  */
  26.  
  27. #include "libavutil/channel_layout.h"
  28. #include "libavutil/common.h"
  29. #include "libavutil/eval.h"
  30. #include "libavutil/float_dsp.h"
  31. #include "libavutil/intreadwrite.h"
  32. #include "libavutil/opt.h"
  33. #include "libavutil/replaygain.h"
  34.  
  35. #include "audio.h"
  36. #include "avfilter.h"
  37. #include "formats.h"
  38. #include "internal.h"
  39. #include "af_volume.h"
  40.  
  41. static const char * const precision_str[] = {
  42.     "fixed", "float", "double"
  43. };
  44.  
  45. static const char *const var_names[] = {
  46.     "n",                   ///< frame number (starting at zero)
  47.     "nb_channels",         ///< number of channels
  48.     "nb_consumed_samples", ///< number of samples consumed by the filter
  49.     "nb_samples",          ///< number of samples in the current frame
  50.     "pos",                 ///< position in the file of the frame
  51.     "pts",                 ///< frame presentation timestamp
  52.     "sample_rate",         ///< sample rate
  53.     "startpts",            ///< PTS at start of stream
  54.     "startt",              ///< time at start of stream
  55.     "t",                   ///< time in the file of the frame
  56.     "tb",                  ///< timebase
  57.     "volume",              ///< last set value
  58.     NULL
  59. };
  60.  
  61. #define OFFSET(x) offsetof(VolumeContext, x)
  62. #define A AV_OPT_FLAG_AUDIO_PARAM
  63. #define F AV_OPT_FLAG_FILTERING_PARAM
  64.  
  65. static const AVOption volume_options[] = {
  66.     { "volume", "set volume adjustment expression",
  67.             OFFSET(volume_expr), AV_OPT_TYPE_STRING, { .str = "1.0" }, .flags = A|F },
  68.     { "precision", "select mathematical precision",
  69.             OFFSET(precision), AV_OPT_TYPE_INT, { .i64 = PRECISION_FLOAT }, PRECISION_FIXED, PRECISION_DOUBLE, A|F, "precision" },
  70.         { "fixed",  "select 8-bit fixed-point",     0, AV_OPT_TYPE_CONST, { .i64 = PRECISION_FIXED  }, INT_MIN, INT_MAX, A|F, "precision" },
  71.         { "float",  "select 32-bit floating-point", 0, AV_OPT_TYPE_CONST, { .i64 = PRECISION_FLOAT  }, INT_MIN, INT_MAX, A|F, "precision" },
  72.         { "double", "select 64-bit floating-point", 0, AV_OPT_TYPE_CONST, { .i64 = PRECISION_DOUBLE }, INT_MIN, INT_MAX, A|F, "precision" },
  73.     { "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" },
  74.          { "once",  "eval volume expression once", 0, AV_OPT_TYPE_CONST, {.i64=EVAL_MODE_ONCE},  .flags = A|F, .unit = "eval" },
  75.          { "frame", "eval volume expression per-frame",                  0, AV_OPT_TYPE_CONST, {.i64=EVAL_MODE_FRAME}, .flags = A|F, .unit = "eval" },
  76.     { "replaygain", "Apply replaygain side data when present",
  77.             OFFSET(replaygain), AV_OPT_TYPE_INT, { .i64 = REPLAYGAIN_DROP }, REPLAYGAIN_DROP, REPLAYGAIN_ALBUM, A, "replaygain" },
  78.         { "drop",   "replaygain side data is dropped", 0, AV_OPT_TYPE_CONST, { .i64 = REPLAYGAIN_DROP   }, 0, 0, A, "replaygain" },
  79.         { "ignore", "replaygain side data is ignored", 0, AV_OPT_TYPE_CONST, { .i64 = REPLAYGAIN_IGNORE }, 0, 0, A, "replaygain" },
  80.         { "track",  "track gain is preferred",         0, AV_OPT_TYPE_CONST, { .i64 = REPLAYGAIN_TRACK  }, 0, 0, A, "replaygain" },
  81.         { "album",  "album gain is preferred",         0, AV_OPT_TYPE_CONST, { .i64 = REPLAYGAIN_ALBUM  }, 0, 0, A, "replaygain" },
  82.     { "replaygain_preamp", "Apply replaygain pre-amplification",
  83.             OFFSET(replaygain_preamp), AV_OPT_TYPE_DOUBLE, { .dbl = 0.0 }, -15.0, 15.0, A },
  84.     { "replaygain_noclip", "Apply replaygain clipping prevention",
  85.             OFFSET(replaygain_noclip), AV_OPT_TYPE_INT, { .i64 = 1 }, 0, 1, A },
  86.     { NULL },
  87. };
  88.  
  89. AVFILTER_DEFINE_CLASS(volume);
  90.  
  91. static int set_expr(AVExpr **pexpr, const char *expr, void *log_ctx)
  92. {
  93.     int ret;
  94.     AVExpr *old = NULL;
  95.  
  96.     if (*pexpr)
  97.         old = *pexpr;
  98.     ret = av_expr_parse(pexpr, expr, var_names,
  99.                         NULL, NULL, NULL, NULL, 0, log_ctx);
  100.     if (ret < 0) {
  101.         av_log(log_ctx, AV_LOG_ERROR,
  102.                "Error when evaluating the volume expression '%s'\n", expr);
  103.         *pexpr = old;
  104.         return ret;
  105.     }
  106.  
  107.     av_expr_free(old);
  108.     return 0;
  109. }
  110.  
  111. static av_cold int init(AVFilterContext *ctx)
  112. {
  113.     VolumeContext *vol = ctx->priv;
  114.  
  115.     vol->fdsp = avpriv_float_dsp_alloc(0);
  116.     if (!vol->fdsp)
  117.         return AVERROR(ENOMEM);
  118.  
  119.     return set_expr(&vol->volume_pexpr, vol->volume_expr, ctx);
  120. }
  121.  
  122. static av_cold void uninit(AVFilterContext *ctx)
  123. {
  124.     VolumeContext *vol = ctx->priv;
  125.     av_expr_free(vol->volume_pexpr);
  126.     av_opt_free(vol);
  127.     av_freep(&vol->fdsp);
  128. }
  129.  
  130. static int query_formats(AVFilterContext *ctx)
  131. {
  132.     VolumeContext *vol = ctx->priv;
  133.     AVFilterFormats *formats = NULL;
  134.     AVFilterChannelLayouts *layouts;
  135.     static const enum AVSampleFormat sample_fmts[][7] = {
  136.         [PRECISION_FIXED] = {
  137.             AV_SAMPLE_FMT_U8,
  138.             AV_SAMPLE_FMT_U8P,
  139.             AV_SAMPLE_FMT_S16,
  140.             AV_SAMPLE_FMT_S16P,
  141.             AV_SAMPLE_FMT_S32,
  142.             AV_SAMPLE_FMT_S32P,
  143.             AV_SAMPLE_FMT_NONE
  144.         },
  145.         [PRECISION_FLOAT] = {
  146.             AV_SAMPLE_FMT_FLT,
  147.             AV_SAMPLE_FMT_FLTP,
  148.             AV_SAMPLE_FMT_NONE
  149.         },
  150.         [PRECISION_DOUBLE] = {
  151.             AV_SAMPLE_FMT_DBL,
  152.             AV_SAMPLE_FMT_DBLP,
  153.             AV_SAMPLE_FMT_NONE
  154.         }
  155.     };
  156.     int ret;
  157.  
  158.     layouts = ff_all_channel_counts();
  159.     if (!layouts)
  160.         return AVERROR(ENOMEM);
  161.     ret = ff_set_common_channel_layouts(ctx, layouts);
  162.     if (ret < 0)
  163.         return ret;
  164.  
  165.     formats = ff_make_format_list(sample_fmts[vol->precision]);
  166.     if (!formats)
  167.         return AVERROR(ENOMEM);
  168.     ret = ff_set_common_formats(ctx, formats);
  169.     if (ret < 0)
  170.         return ret;
  171.  
  172.     formats = ff_all_samplerates();
  173.     if (!formats)
  174.         return AVERROR(ENOMEM);
  175.     return ff_set_common_samplerates(ctx, formats);
  176. }
  177.  
  178. static inline void scale_samples_u8(uint8_t *dst, const uint8_t *src,
  179.                                     int nb_samples, int volume)
  180. {
  181.     int i;
  182.     for (i = 0; i < nb_samples; i++)
  183.         dst[i] = av_clip_uint8(((((int64_t)src[i] - 128) * volume + 128) >> 8) + 128);
  184. }
  185.  
  186. static inline void scale_samples_u8_small(uint8_t *dst, const uint8_t *src,
  187.                                           int nb_samples, int volume)
  188. {
  189.     int i;
  190.     for (i = 0; i < nb_samples; i++)
  191.         dst[i] = av_clip_uint8((((src[i] - 128) * volume + 128) >> 8) + 128);
  192. }
  193.  
  194. static inline void scale_samples_s16(uint8_t *dst, const uint8_t *src,
  195.                                      int nb_samples, int volume)
  196. {
  197.     int i;
  198.     int16_t *smp_dst       = (int16_t *)dst;
  199.     const int16_t *smp_src = (const int16_t *)src;
  200.     for (i = 0; i < nb_samples; i++)
  201.         smp_dst[i] = av_clip_int16(((int64_t)smp_src[i] * volume + 128) >> 8);
  202. }
  203.  
  204. static inline void scale_samples_s16_small(uint8_t *dst, const uint8_t *src,
  205.                                            int nb_samples, int volume)
  206. {
  207.     int i;
  208.     int16_t *smp_dst       = (int16_t *)dst;
  209.     const int16_t *smp_src = (const int16_t *)src;
  210.     for (i = 0; i < nb_samples; i++)
  211.         smp_dst[i] = av_clip_int16((smp_src[i] * volume + 128) >> 8);
  212. }
  213.  
  214. static inline void scale_samples_s32(uint8_t *dst, const uint8_t *src,
  215.                                      int nb_samples, int volume)
  216. {
  217.     int i;
  218.     int32_t *smp_dst       = (int32_t *)dst;
  219.     const int32_t *smp_src = (const int32_t *)src;
  220.     for (i = 0; i < nb_samples; i++)
  221.         smp_dst[i] = av_clipl_int32((((int64_t)smp_src[i] * volume + 128) >> 8));
  222. }
  223.  
  224. static av_cold void volume_init(VolumeContext *vol)
  225. {
  226.     vol->samples_align = 1;
  227.  
  228.     switch (av_get_packed_sample_fmt(vol->sample_fmt)) {
  229.     case AV_SAMPLE_FMT_U8:
  230.         if (vol->volume_i < 0x1000000)
  231.             vol->scale_samples = scale_samples_u8_small;
  232.         else
  233.             vol->scale_samples = scale_samples_u8;
  234.         break;
  235.     case AV_SAMPLE_FMT_S16:
  236.         if (vol->volume_i < 0x10000)
  237.             vol->scale_samples = scale_samples_s16_small;
  238.         else
  239.             vol->scale_samples = scale_samples_s16;
  240.         break;
  241.     case AV_SAMPLE_FMT_S32:
  242.         vol->scale_samples = scale_samples_s32;
  243.         break;
  244.     case AV_SAMPLE_FMT_FLT:
  245.         vol->samples_align = 4;
  246.         break;
  247.     case AV_SAMPLE_FMT_DBL:
  248.         vol->samples_align = 8;
  249.         break;
  250.     }
  251.  
  252.     if (ARCH_X86)
  253.         ff_volume_init_x86(vol);
  254. }
  255.  
  256. static int set_volume(AVFilterContext *ctx)
  257. {
  258.     VolumeContext *vol = ctx->priv;
  259.  
  260.     vol->volume = av_expr_eval(vol->volume_pexpr, vol->var_values, NULL);
  261.     if (isnan(vol->volume)) {
  262.         if (vol->eval_mode == EVAL_MODE_ONCE) {
  263.             av_log(ctx, AV_LOG_ERROR, "Invalid value NaN for volume\n");
  264.             return AVERROR(EINVAL);
  265.         } else {
  266.             av_log(ctx, AV_LOG_WARNING, "Invalid value NaN for volume, setting to 0\n");
  267.             vol->volume = 0;
  268.         }
  269.     }
  270.     vol->var_values[VAR_VOLUME] = vol->volume;
  271.  
  272.     av_log(ctx, AV_LOG_VERBOSE, "n:%f t:%f pts:%f precision:%s ",
  273.            vol->var_values[VAR_N], vol->var_values[VAR_T], vol->var_values[VAR_PTS],
  274.            precision_str[vol->precision]);
  275.  
  276.     if (vol->precision == PRECISION_FIXED) {
  277.         vol->volume_i = (int)(vol->volume * 256 + 0.5);
  278.         vol->volume   = vol->volume_i / 256.0;
  279.         av_log(ctx, AV_LOG_VERBOSE, "volume_i:%d/255 ", vol->volume_i);
  280.     }
  281.     av_log(ctx, AV_LOG_VERBOSE, "volume:%f volume_dB:%f\n",
  282.            vol->volume, 20.0*log(vol->volume)/M_LN10);
  283.  
  284.     volume_init(vol);
  285.     return 0;
  286. }
  287.  
  288. static int config_output(AVFilterLink *outlink)
  289. {
  290.     AVFilterContext *ctx = outlink->src;
  291.     VolumeContext *vol   = ctx->priv;
  292.     AVFilterLink *inlink = ctx->inputs[0];
  293.  
  294.     vol->sample_fmt = inlink->format;
  295.     vol->channels   = inlink->channels;
  296.     vol->planes     = av_sample_fmt_is_planar(inlink->format) ? vol->channels : 1;
  297.  
  298.     vol->var_values[VAR_N] =
  299.     vol->var_values[VAR_NB_CONSUMED_SAMPLES] =
  300.     vol->var_values[VAR_NB_SAMPLES] =
  301.     vol->var_values[VAR_POS] =
  302.     vol->var_values[VAR_PTS] =
  303.     vol->var_values[VAR_STARTPTS] =
  304.     vol->var_values[VAR_STARTT] =
  305.     vol->var_values[VAR_T] =
  306.     vol->var_values[VAR_VOLUME] = NAN;
  307.  
  308.     vol->var_values[VAR_NB_CHANNELS] = inlink->channels;
  309.     vol->var_values[VAR_TB]          = av_q2d(inlink->time_base);
  310.     vol->var_values[VAR_SAMPLE_RATE] = inlink->sample_rate;
  311.  
  312.     av_log(inlink->src, AV_LOG_VERBOSE, "tb:%f sample_rate:%f nb_channels:%f\n",
  313.            vol->var_values[VAR_TB],
  314.            vol->var_values[VAR_SAMPLE_RATE],
  315.            vol->var_values[VAR_NB_CHANNELS]);
  316.  
  317.     return set_volume(ctx);
  318. }
  319.  
  320. static int process_command(AVFilterContext *ctx, const char *cmd, const char *args,
  321.                            char *res, int res_len, int flags)
  322. {
  323.     VolumeContext *vol = ctx->priv;
  324.     int ret = AVERROR(ENOSYS);
  325.  
  326.     if (!strcmp(cmd, "volume")) {
  327.         if ((ret = set_expr(&vol->volume_pexpr, args, ctx)) < 0)
  328.             return ret;
  329.         if (vol->eval_mode == EVAL_MODE_ONCE)
  330.             set_volume(ctx);
  331.     }
  332.  
  333.     return ret;
  334. }
  335.  
  336. #define D2TS(d)  (isnan(d) ? AV_NOPTS_VALUE : (int64_t)(d))
  337. #define TS2D(ts) ((ts) == AV_NOPTS_VALUE ? NAN : (double)(ts))
  338. #define TS2T(ts, tb) ((ts) == AV_NOPTS_VALUE ? NAN : (double)(ts)*av_q2d(tb))
  339.  
  340. static int filter_frame(AVFilterLink *inlink, AVFrame *buf)
  341. {
  342.     AVFilterContext *ctx = inlink->dst;
  343.     VolumeContext *vol    = inlink->dst->priv;
  344.     AVFilterLink *outlink = inlink->dst->outputs[0];
  345.     int nb_samples        = buf->nb_samples;
  346.     AVFrame *out_buf;
  347.     int64_t pos;
  348.     AVFrameSideData *sd = av_frame_get_side_data(buf, AV_FRAME_DATA_REPLAYGAIN);
  349.     int ret;
  350.  
  351.     if (sd && vol->replaygain != REPLAYGAIN_IGNORE) {
  352.         if (vol->replaygain != REPLAYGAIN_DROP) {
  353.             AVReplayGain *replaygain = (AVReplayGain*)sd->data;
  354.             int32_t gain  = 100000;
  355.             uint32_t peak = 100000;
  356.             float g, p;
  357.  
  358.             if (vol->replaygain == REPLAYGAIN_TRACK &&
  359.                 replaygain->track_gain != INT32_MIN) {
  360.                 gain = replaygain->track_gain;
  361.  
  362.                 if (replaygain->track_peak != 0)
  363.                     peak = replaygain->track_peak;
  364.             } else if (replaygain->album_gain != INT32_MIN) {
  365.                 gain = replaygain->album_gain;
  366.  
  367.                 if (replaygain->album_peak != 0)
  368.                     peak = replaygain->album_peak;
  369.             } else {
  370.                 av_log(inlink->dst, AV_LOG_WARNING, "Both ReplayGain gain "
  371.                        "values are unknown.\n");
  372.             }
  373.             g = gain / 100000.0f;
  374.             p = peak / 100000.0f;
  375.  
  376.             av_log(inlink->dst, AV_LOG_VERBOSE,
  377.                    "Using gain %f dB from replaygain side data.\n", g);
  378.  
  379.             vol->volume   = pow(10, (g + vol->replaygain_preamp) / 20);
  380.             if (vol->replaygain_noclip)
  381.                 vol->volume = FFMIN(vol->volume, 1.0 / p);
  382.             vol->volume_i = (int)(vol->volume * 256 + 0.5);
  383.  
  384.             volume_init(vol);
  385.         }
  386.         av_frame_remove_side_data(buf, AV_FRAME_DATA_REPLAYGAIN);
  387.     }
  388.  
  389.     if (isnan(vol->var_values[VAR_STARTPTS])) {
  390.         vol->var_values[VAR_STARTPTS] = TS2D(buf->pts);
  391.         vol->var_values[VAR_STARTT  ] = TS2T(buf->pts, inlink->time_base);
  392.     }
  393.     vol->var_values[VAR_PTS] = TS2D(buf->pts);
  394.     vol->var_values[VAR_T  ] = TS2T(buf->pts, inlink->time_base);
  395.     vol->var_values[VAR_N  ] = inlink->frame_count;
  396.  
  397.     pos = av_frame_get_pkt_pos(buf);
  398.     vol->var_values[VAR_POS] = pos == -1 ? NAN : pos;
  399.     if (vol->eval_mode == EVAL_MODE_FRAME)
  400.         set_volume(ctx);
  401.  
  402.     if (vol->volume == 1.0 || vol->volume_i == 256) {
  403.         out_buf = buf;
  404.         goto end;
  405.     }
  406.  
  407.     /* do volume scaling in-place if input buffer is writable */
  408.     if (av_frame_is_writable(buf)
  409.             && (vol->precision != PRECISION_FIXED || vol->volume_i > 0)) {
  410.         out_buf = buf;
  411.     } else {
  412.         out_buf = ff_get_audio_buffer(inlink, nb_samples);
  413.         if (!out_buf)
  414.             return AVERROR(ENOMEM);
  415.         ret = av_frame_copy_props(out_buf, buf);
  416.         if (ret < 0) {
  417.             av_frame_free(&out_buf);
  418.             av_frame_free(&buf);
  419.             return ret;
  420.         }
  421.     }
  422.  
  423.     if (vol->precision != PRECISION_FIXED || vol->volume_i > 0) {
  424.         int p, plane_samples;
  425.  
  426.         if (av_sample_fmt_is_planar(buf->format))
  427.             plane_samples = FFALIGN(nb_samples, vol->samples_align);
  428.         else
  429.             plane_samples = FFALIGN(nb_samples * vol->channels, vol->samples_align);
  430.  
  431.         if (vol->precision == PRECISION_FIXED) {
  432.             for (p = 0; p < vol->planes; p++) {
  433.                 vol->scale_samples(out_buf->extended_data[p],
  434.                                    buf->extended_data[p], plane_samples,
  435.                                    vol->volume_i);
  436.             }
  437.         } else if (av_get_packed_sample_fmt(vol->sample_fmt) == AV_SAMPLE_FMT_FLT) {
  438.             for (p = 0; p < vol->planes; p++) {
  439.                 vol->fdsp->vector_fmul_scalar((float *)out_buf->extended_data[p],
  440.                                              (const float *)buf->extended_data[p],
  441.                                              vol->volume, plane_samples);
  442.             }
  443.         } else {
  444.             for (p = 0; p < vol->planes; p++) {
  445.                 vol->fdsp->vector_dmul_scalar((double *)out_buf->extended_data[p],
  446.                                              (const double *)buf->extended_data[p],
  447.                                              vol->volume, plane_samples);
  448.             }
  449.         }
  450.     }
  451.  
  452.     emms_c();
  453.  
  454.     if (buf != out_buf)
  455.         av_frame_free(&buf);
  456.  
  457. end:
  458.     vol->var_values[VAR_NB_CONSUMED_SAMPLES] += out_buf->nb_samples;
  459.     return ff_filter_frame(outlink, out_buf);
  460. }
  461.  
  462. static const AVFilterPad avfilter_af_volume_inputs[] = {
  463.     {
  464.         .name           = "default",
  465.         .type           = AVMEDIA_TYPE_AUDIO,
  466.         .filter_frame   = filter_frame,
  467.     },
  468.     { NULL }
  469. };
  470.  
  471. static const AVFilterPad avfilter_af_volume_outputs[] = {
  472.     {
  473.         .name         = "default",
  474.         .type         = AVMEDIA_TYPE_AUDIO,
  475.         .config_props = config_output,
  476.     },
  477.     { NULL }
  478. };
  479.  
  480. AVFilter ff_af_volume = {
  481.     .name           = "volume",
  482.     .description    = NULL_IF_CONFIG_SMALL("Change input volume."),
  483.     .query_formats  = query_formats,
  484.     .priv_size      = sizeof(VolumeContext),
  485.     .priv_class     = &volume_class,
  486.     .init           = init,
  487.     .uninit         = uninit,
  488.     .inputs         = avfilter_af_volume_inputs,
  489.     .outputs        = avfilter_af_volume_outputs,
  490.     .flags          = AVFILTER_FLAG_SUPPORT_TIMELINE_GENERIC,
  491.     .process_command = process_command,
  492. };
  493.