/* * 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/ffmath.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|F, "replaygain" },        { "drop",   "replaygain side data is dropped", 0, AV_OPT_TYPE_CONST, { .i64 = REPLAYGAIN_DROP   }, 0, 0, A|F, "replaygain" },        { "ignore", "replaygain side data is ignored", 0, AV_OPT_TYPE_CONST, { .i64 = REPLAYGAIN_IGNORE }, 0, 0, A|F, "replaygain" },        { "track",  "track gain is preferred",         0, AV_OPT_TYPE_CONST, { .i64 = REPLAYGAIN_TRACK  }, 0, 0, A|F, "replaygain" },        { "album",  "album gain is preferred",         0, AV_OPT_TYPE_CONST, { .i64 = REPLAYGAIN_ALBUM  }, 0, 0, A|F, "replaygain" },    { "replaygain_preamp", "Apply replaygain pre-amplification",            OFFSET(replaygain_preamp), AV_OPT_TYPE_DOUBLE, { .dbl = 0.0 }, -15.0, 15.0, A|F },    { "replaygain_noclip", "Apply replaygain clipping prevention",            OFFSET(replaygain_noclip), AV_OPT_TYPE_BOOL, { .i64 = 1 }, 0, 1, A|F },    { 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        }    };    int ret;     layouts = ff_all_channel_counts();    if (!layouts)        return AVERROR(ENOMEM);    ret = ff_set_common_channel_layouts(ctx, layouts);    if (ret < 0)        return ret;     formats = ff_make_format_list(sample_fmts[vol->precision]);    if (!formats)        return AVERROR(ENOMEM);    ret = ff_set_common_formats(ctx, formats);    if (ret < 0)        return ret;     formats = ff_all_samplerates();    if (!formats)        return AVERROR(ENOMEM);    return ff_set_common_samplerates(ctx, formats);} 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*log10(vol->volume));     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   = ff_exp10((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_out;     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)            && (vol->precision != PRECISION_FIXED || vol->volume_i > 0)) {        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,};