/* * 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 #if FF_API_FRAME_PKT "pos", ///< position in the file of the frame #endif "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 #define T AV_OPT_FLAG_RUNTIME_PARAM static const AVOption volume_options[] = { { "volume", "set volume adjustment expression", OFFSET(volume_expr), AV_OPT_TYPE_STRING, { .str = "1.0" }, .flags = A|F|T }, { "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; 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 = ff_set_common_all_channel_counts(ctx); if (ret < 0) return ret; ret = ff_set_common_formats_from_list(ctx, sample_fmts[vol->precision]); if (ret < 0) return ret; return ff_set_common_all_samplerates(ctx); } 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); #endif } 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->ch_layout.nb_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] = #if FF_API_FRAME_PKT vol->var_values[VAR_POS] = #endif 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->ch_layout.nb_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; } 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; 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; #if FF_API_FRAME_PKT FF_DISABLE_DEPRECATION_WARNINGS { int64_t pos; pos = buf->pkt_pos; vol->var_values[VAR_POS] = pos == -1 ? NAN : pos; } FF_ENABLE_DEPRECATION_WARNINGS #endif 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(outlink, nb_samples); if (!out_buf) { av_frame_free(&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, }, }; static const AVFilterPad avfilter_af_volume_outputs[] = { { .name = "default", .type = AVMEDIA_TYPE_AUDIO, .config_props = config_output, }, }; const AVFilter ff_af_volume = { .name = "volume", .description = NULL_IF_CONFIG_SMALL("Change input volume."), .priv_size = sizeof(VolumeContext), .priv_class = &volume_class, .init = init, .uninit = uninit, FILTER_INPUTS(avfilter_af_volume_inputs), FILTER_OUTPUTS(avfilter_af_volume_outputs), FILTER_QUERY_FUNC(query_formats), .flags = AVFILTER_FLAG_SUPPORT_TIMELINE_GENERIC, .process_command = process_command, };