LCOV - code coverage report
Current view: top level - src/libavfilter - af_chorus.c (source / functions) Hit Total Coverage
Test: coverage.info Lines: 139 165 84.2 %
Date: 2017-08-17 10:06:07 Functions: 8 8 100.0 %

          Line data    Source code
       1             : /*
       2             :  * Copyright (c) 1998 Juergen Mueller And Sundry Contributors
       3             :  * This source code is freely redistributable and may be used for
       4             :  * any purpose.  This copyright notice must be maintained.
       5             :  * Juergen Mueller And Sundry Contributors are not responsible for
       6             :  * the consequences of using this software.
       7             :  *
       8             :  * Copyright (c) 2015 Paul B Mahol
       9             :  *
      10             :  * This file is part of FFmpeg.
      11             :  *
      12             :  * FFmpeg is free software; you can redistribute it and/or
      13             :  * modify it under the terms of the GNU Lesser General Public
      14             :  * License as published by the Free Software Foundation; either
      15             :  * version 2.1 of the License, or (at your option) any later version.
      16             :  *
      17             :  * FFmpeg is distributed in the hope that it will be useful,
      18             :  * but WITHOUT ANY WARRANTY; without even the implied warranty of
      19             :  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
      20             :  * Lesser General Public License for more details.
      21             :  *
      22             :  * You should have received a copy of the GNU Lesser General Public
      23             :  * License along with FFmpeg; if not, write to the Free Software
      24             :  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
      25             :  */
      26             : 
      27             : /**
      28             :  * @file
      29             :  * chorus audio filter
      30             :  */
      31             : 
      32             : #include "libavutil/avstring.h"
      33             : #include "libavutil/opt.h"
      34             : #include "audio.h"
      35             : #include "avfilter.h"
      36             : #include "internal.h"
      37             : #include "generate_wave_table.h"
      38             : 
      39             : typedef struct ChorusContext {
      40             :     const AVClass *class;
      41             :     float in_gain, out_gain;
      42             :     char *delays_str;
      43             :     char *decays_str;
      44             :     char *speeds_str;
      45             :     char *depths_str;
      46             :     float *delays;
      47             :     float *decays;
      48             :     float *speeds;
      49             :     float *depths;
      50             :     uint8_t **chorusbuf;
      51             :     int **phase;
      52             :     int *length;
      53             :     int32_t **lookup_table;
      54             :     int *counter;
      55             :     int num_chorus;
      56             :     int max_samples;
      57             :     int channels;
      58             :     int modulation;
      59             :     int fade_out;
      60             :     int64_t next_pts;
      61             : } ChorusContext;
      62             : 
      63             : #define OFFSET(x) offsetof(ChorusContext, x)
      64             : #define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
      65             : 
      66             : static const AVOption chorus_options[] = {
      67             :     { "in_gain",  "set input gain",  OFFSET(in_gain),    AV_OPT_TYPE_FLOAT,  {.dbl=.4}, 0, 1, A },
      68             :     { "out_gain", "set output gain", OFFSET(out_gain),   AV_OPT_TYPE_FLOAT,  {.dbl=.4}, 0, 1, A },
      69             :     { "delays",   "set delays",      OFFSET(delays_str), AV_OPT_TYPE_STRING, {.str=NULL}, 0, 0, A },
      70             :     { "decays",   "set decays",      OFFSET(decays_str), AV_OPT_TYPE_STRING, {.str=NULL}, 0, 0, A },
      71             :     { "speeds",   "set speeds",      OFFSET(speeds_str), AV_OPT_TYPE_STRING, {.str=NULL}, 0, 0, A },
      72             :     { "depths",   "set depths",      OFFSET(depths_str), AV_OPT_TYPE_STRING, {.str=NULL}, 0, 0, A },
      73             :     { NULL }
      74             : };
      75             : 
      76             : AVFILTER_DEFINE_CLASS(chorus);
      77             : 
      78           4 : static void count_items(char *item_str, int *nb_items)
      79             : {
      80             :     char *p;
      81             : 
      82           4 :     *nb_items = 1;
      83          29 :     for (p = item_str; *p; p++) {
      84          25 :         if (*p == '|')
      85           0 :             (*nb_items)++;
      86             :     }
      87             : 
      88           4 : }
      89             : 
      90           4 : static void fill_items(char *item_str, int *nb_items, float *items)
      91             : {
      92           4 :     char *p, *saveptr = NULL;
      93           4 :     int i, new_nb_items = 0;
      94             : 
      95           4 :     p = item_str;
      96           8 :     for (i = 0; i < *nb_items; i++) {
      97           4 :         char *tstr = av_strtok(p, "|", &saveptr);
      98           4 :         p = NULL;
      99           4 :         if (tstr)
     100           4 :             new_nb_items += sscanf(tstr, "%f", &items[new_nb_items]) == 1;
     101             :     }
     102             : 
     103           4 :     *nb_items = new_nb_items;
     104           4 : }
     105             : 
     106           1 : static av_cold int init(AVFilterContext *ctx)
     107             : {
     108           1 :     ChorusContext *s = ctx->priv;
     109             :     int nb_delays, nb_decays, nb_speeds, nb_depths;
     110             : 
     111           1 :     if (!s->delays_str || !s->decays_str || !s->speeds_str || !s->depths_str) {
     112           0 :         av_log(ctx, AV_LOG_ERROR, "Both delays & decays & speeds & depths must be set.\n");
     113           0 :         return AVERROR(EINVAL);
     114             :     }
     115             : 
     116           1 :     count_items(s->delays_str, &nb_delays);
     117           1 :     count_items(s->decays_str, &nb_decays);
     118           1 :     count_items(s->speeds_str, &nb_speeds);
     119           1 :     count_items(s->depths_str, &nb_depths);
     120             : 
     121           1 :     s->delays = av_realloc_f(s->delays, nb_delays, sizeof(*s->delays));
     122           1 :     s->decays = av_realloc_f(s->decays, nb_decays, sizeof(*s->decays));
     123           1 :     s->speeds = av_realloc_f(s->speeds, nb_speeds, sizeof(*s->speeds));
     124           1 :     s->depths = av_realloc_f(s->depths, nb_depths, sizeof(*s->depths));
     125             : 
     126           1 :     if (!s->delays || !s->decays || !s->speeds || !s->depths)
     127           0 :         return AVERROR(ENOMEM);
     128             : 
     129           1 :     fill_items(s->delays_str, &nb_delays, s->delays);
     130           1 :     fill_items(s->decays_str, &nb_decays, s->decays);
     131           1 :     fill_items(s->speeds_str, &nb_speeds, s->speeds);
     132           1 :     fill_items(s->depths_str, &nb_depths, s->depths);
     133             : 
     134           1 :     if (nb_delays != nb_decays && nb_delays != nb_speeds && nb_delays != nb_depths) {
     135           0 :         av_log(ctx, AV_LOG_ERROR, "Number of delays & decays & speeds & depths given must be same.\n");
     136           0 :         return AVERROR(EINVAL);
     137             :     }
     138             : 
     139           1 :     s->num_chorus = nb_delays;
     140             : 
     141           1 :     if (s->num_chorus < 1) {
     142           0 :         av_log(ctx, AV_LOG_ERROR, "At least one delay & decay & speed & depth must be set.\n");
     143           0 :         return AVERROR(EINVAL);
     144             :     }
     145             : 
     146           1 :     s->length = av_calloc(s->num_chorus, sizeof(*s->length));
     147           1 :     s->lookup_table = av_calloc(s->num_chorus, sizeof(*s->lookup_table));
     148             : 
     149           1 :     if (!s->length || !s->lookup_table)
     150           0 :         return AVERROR(ENOMEM);
     151             : 
     152           1 :     s->next_pts = AV_NOPTS_VALUE;
     153             : 
     154           1 :     return 0;
     155             : }
     156             : 
     157           1 : static int query_formats(AVFilterContext *ctx)
     158             : {
     159             :     AVFilterFormats *formats;
     160             :     AVFilterChannelLayouts *layouts;
     161             :     static const enum AVSampleFormat sample_fmts[] = {
     162             :         AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_NONE
     163             :     };
     164             :     int ret;
     165             : 
     166           1 :     layouts = ff_all_channel_counts();
     167           1 :     if (!layouts)
     168           0 :         return AVERROR(ENOMEM);
     169           1 :     ret = ff_set_common_channel_layouts(ctx, layouts);
     170           1 :     if (ret < 0)
     171           0 :         return ret;
     172             : 
     173           1 :     formats = ff_make_format_list(sample_fmts);
     174           1 :     if (!formats)
     175           0 :         return AVERROR(ENOMEM);
     176           1 :     ret = ff_set_common_formats(ctx, formats);
     177           1 :     if (ret < 0)
     178           0 :         return ret;
     179             : 
     180           1 :     formats = ff_all_samplerates();
     181           1 :     if (!formats)
     182           0 :         return AVERROR(ENOMEM);
     183           1 :     return ff_set_common_samplerates(ctx, formats);
     184             : }
     185             : 
     186           1 : static int config_output(AVFilterLink *outlink)
     187             : {
     188           1 :     AVFilterContext *ctx = outlink->src;
     189           1 :     ChorusContext *s = ctx->priv;
     190           1 :     float sum_in_volume = 1.0;
     191             :     int n;
     192             : 
     193           1 :     s->channels = outlink->channels;
     194             : 
     195           2 :     for (n = 0; n < s->num_chorus; n++) {
     196           1 :         int samples = (int) ((s->delays[n] + s->depths[n]) * outlink->sample_rate / 1000.0);
     197           1 :         int depth_samples = (int) (s->depths[n] * outlink->sample_rate / 1000.0);
     198             : 
     199           1 :         s->length[n] = outlink->sample_rate / s->speeds[n];
     200             : 
     201           1 :         s->lookup_table[n] = av_malloc(sizeof(int32_t) * s->length[n]);
     202           1 :         if (!s->lookup_table[n])
     203           0 :             return AVERROR(ENOMEM);
     204             : 
     205           2 :         ff_generate_wave_table(WAVE_SIN, AV_SAMPLE_FMT_S32, s->lookup_table[n],
     206           1 :                                s->length[n], 0., depth_samples, 0);
     207           1 :         s->max_samples = FFMAX(s->max_samples, samples);
     208             :     }
     209             : 
     210           2 :     for (n = 0; n < s->num_chorus; n++)
     211           1 :         sum_in_volume += s->decays[n];
     212             : 
     213           1 :     if (s->in_gain * (sum_in_volume) > 1.0 / s->out_gain)
     214           0 :         av_log(ctx, AV_LOG_WARNING, "output gain can cause saturation or clipping of output\n");
     215             : 
     216           1 :     s->counter = av_calloc(outlink->channels, sizeof(*s->counter));
     217           1 :     if (!s->counter)
     218           0 :         return AVERROR(ENOMEM);
     219             : 
     220           1 :     s->phase = av_calloc(outlink->channels, sizeof(*s->phase));
     221           1 :     if (!s->phase)
     222           0 :         return AVERROR(ENOMEM);
     223             : 
     224           2 :     for (n = 0; n < outlink->channels; n++) {
     225           1 :         s->phase[n] = av_calloc(s->num_chorus, sizeof(int));
     226           1 :         if (!s->phase[n])
     227           0 :             return AVERROR(ENOMEM);
     228             :     }
     229             : 
     230           1 :     s->fade_out = s->max_samples;
     231             : 
     232           1 :     return av_samples_alloc_array_and_samples(&s->chorusbuf, NULL,
     233             :                                               outlink->channels,
     234             :                                               s->max_samples,
     235           1 :                                               outlink->format, 0);
     236             : }
     237             : 
     238             : #define MOD(a, b) (((a) >= (b)) ? (a) - (b) : (a))
     239             : 
     240          11 : static int filter_frame(AVFilterLink *inlink, AVFrame *frame)
     241             : {
     242          11 :     AVFilterContext *ctx = inlink->dst;
     243          11 :     ChorusContext *s = ctx->priv;
     244             :     AVFrame *out_frame;
     245             :     int c, i, n;
     246             : 
     247          11 :     if (av_frame_is_writable(frame)) {
     248          11 :         out_frame = frame;
     249             :     } else {
     250           0 :         out_frame = ff_get_audio_buffer(inlink, frame->nb_samples);
     251           0 :         if (!out_frame) {
     252           0 :             av_frame_free(&frame);
     253           0 :             return AVERROR(ENOMEM);
     254             :         }
     255           0 :         av_frame_copy_props(out_frame, frame);
     256             :     }
     257             : 
     258          22 :     for (c = 0; c < inlink->channels; c++) {
     259          11 :         const float *src = (const float *)frame->extended_data[c];
     260          11 :         float *dst = (float *)out_frame->extended_data[c];
     261          11 :         float *chorusbuf = (float *)s->chorusbuf[c];
     262          11 :         int *phase = s->phase[c];
     263             : 
     264       21946 :         for (i = 0; i < frame->nb_samples; i++) {
     265       21935 :             float out, in = src[i];
     266             : 
     267       21935 :             out = in * s->in_gain;
     268             : 
     269       43870 :             for (n = 0; n < s->num_chorus; n++) {
     270       43870 :                 out += chorusbuf[MOD(s->max_samples + s->counter[c] -
     271             :                                      s->lookup_table[n][phase[n]],
     272       21935 :                                      s->max_samples)] * s->decays[n];
     273       21935 :                 phase[n] = MOD(phase[n] + 1, s->length[n]);
     274             :             }
     275             : 
     276       21935 :             out *= s->out_gain;
     277             : 
     278       21935 :             dst[i] = out;
     279             : 
     280       21935 :             chorusbuf[s->counter[c]] = in;
     281       21935 :             s->counter[c] = MOD(s->counter[c] + 1, s->max_samples);
     282             :         }
     283             :     }
     284             : 
     285          11 :     s->next_pts = frame->pts + av_rescale_q(frame->nb_samples, (AVRational){1, inlink->sample_rate}, inlink->time_base);
     286             : 
     287          11 :     if (frame != out_frame)
     288           0 :         av_frame_free(&frame);
     289             : 
     290          11 :     return ff_filter_frame(ctx->outputs[0], out_frame);
     291             : }
     292             : 
     293          10 : static int request_frame(AVFilterLink *outlink)
     294             : {
     295          10 :     AVFilterContext *ctx = outlink->src;
     296          10 :     ChorusContext *s = ctx->priv;
     297             :     int ret;
     298             : 
     299          10 :     ret = ff_request_frame(ctx->inputs[0]);
     300             : 
     301          10 :     if (ret == AVERROR_EOF && !ctx->is_disabled && s->fade_out) {
     302           1 :         int nb_samples = FFMIN(s->fade_out, 2048);
     303             :         AVFrame *frame;
     304             : 
     305           1 :         frame = ff_get_audio_buffer(outlink, nb_samples);
     306           1 :         if (!frame)
     307           0 :             return AVERROR(ENOMEM);
     308           1 :         s->fade_out -= nb_samples;
     309             : 
     310           1 :         av_samples_set_silence(frame->extended_data, 0,
     311             :                                frame->nb_samples,
     312             :                                outlink->channels,
     313           1 :                                frame->format);
     314             : 
     315           1 :         frame->pts = s->next_pts;
     316           1 :         if (s->next_pts != AV_NOPTS_VALUE)
     317           1 :             s->next_pts += av_rescale_q(nb_samples, (AVRational){1, outlink->sample_rate}, outlink->time_base);
     318             : 
     319           1 :         ret = filter_frame(ctx->inputs[0], frame);
     320             :     }
     321             : 
     322          10 :     return ret;
     323             : }
     324             : 
     325           1 : static av_cold void uninit(AVFilterContext *ctx)
     326             : {
     327           1 :     ChorusContext *s = ctx->priv;
     328             :     int n;
     329             : 
     330           1 :     av_freep(&s->delays);
     331           1 :     av_freep(&s->decays);
     332           1 :     av_freep(&s->speeds);
     333           1 :     av_freep(&s->depths);
     334             : 
     335           1 :     if (s->chorusbuf)
     336           1 :         av_freep(&s->chorusbuf[0]);
     337           1 :     av_freep(&s->chorusbuf);
     338             : 
     339           1 :     if (s->phase)
     340           2 :         for (n = 0; n < s->channels; n++)
     341           1 :             av_freep(&s->phase[n]);
     342           1 :     av_freep(&s->phase);
     343             : 
     344           1 :     av_freep(&s->counter);
     345           1 :     av_freep(&s->length);
     346             : 
     347           1 :     if (s->lookup_table)
     348           2 :         for (n = 0; n < s->num_chorus; n++)
     349           1 :             av_freep(&s->lookup_table[n]);
     350           1 :     av_freep(&s->lookup_table);
     351           1 : }
     352             : 
     353             : static const AVFilterPad chorus_inputs[] = {
     354             :     {
     355             :         .name         = "default",
     356             :         .type         = AVMEDIA_TYPE_AUDIO,
     357             :         .filter_frame = filter_frame,
     358             :     },
     359             :     { NULL }
     360             : };
     361             : 
     362             : static const AVFilterPad chorus_outputs[] = {
     363             :     {
     364             :         .name          = "default",
     365             :         .type          = AVMEDIA_TYPE_AUDIO,
     366             :         .request_frame = request_frame,
     367             :         .config_props  = config_output,
     368             :     },
     369             :     { NULL }
     370             : };
     371             : 
     372             : AVFilter ff_af_chorus = {
     373             :     .name          = "chorus",
     374             :     .description   = NULL_IF_CONFIG_SMALL("Add a chorus effect to the audio."),
     375             :     .query_formats = query_formats,
     376             :     .priv_size     = sizeof(ChorusContext),
     377             :     .priv_class    = &chorus_class,
     378             :     .init          = init,
     379             :     .uninit        = uninit,
     380             :     .inputs        = chorus_inputs,
     381             :     .outputs       = chorus_outputs,
     382             : };

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