/*
* Copyright (c) 2011 Stefano Sabatini
+ * Copyright (c) 2012 Justin Ruggles <justin.ruggles@gmail.com>
*
* This file is part of FFmpeg.
*
/**
* @file
* audio volume filter
- * based on ffmpeg.c code
*/
-#include "libavutil/channel_layout.h"
+#include "libavutil/audioconvert.h"
+#include "libavutil/common.h"
#include "libavutil/eval.h"
+#include "libavutil/float_dsp.h"
+#include "libavutil/opt.h"
#include "audio.h"
#include "avfilter.h"
#include "formats.h"
+#include "internal.h"
+#include "af_volume.h"
-typedef struct {
- double volume;
- int volume_i;
-} VolumeContext;
+static const char *precision_str[] = {
+ "fixed", "float", "double"
+};
+
+#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",
+ OFFSET(volume), AV_OPT_TYPE_DOUBLE, { .dbl = 1.0 }, 0, 0x7fffff, 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" },
+ { NULL },
+};
+
+AVFILTER_DEFINE_CLASS(volume);
static av_cold int init(AVFilterContext *ctx, const char *args)
{
VolumeContext *vol = ctx->priv;
- char *tail;
- int ret = 0;
-
- vol->volume = 1.0;
-
- if (args) {
- /* parse the number as a decimal number */
- double d = strtod(args, &tail);
-
- if (*tail) {
- if (!strcmp(tail, "dB")) {
- /* consider the argument an adjustement in decibels */
- d = pow(10, d/20);
- } else {
- /* parse the argument as an expression */
- ret = av_expr_parse_and_eval(&d, args, NULL, NULL,
- NULL, NULL, NULL, NULL,
- NULL, 0, ctx);
- }
- }
+ static const char *shorthand[] = { "volume", "precision", NULL };
+ int ret;
- if (ret < 0) {
- av_log(ctx, AV_LOG_ERROR,
- "Invalid volume argument '%s'\n", args);
- return AVERROR(EINVAL);
- }
+ vol->class = &volume_class;
+ av_opt_set_defaults(vol);
- if (d < 0 || d > 65536) { /* 65536 = INT_MIN / (128 * 256) */
- av_log(ctx, AV_LOG_ERROR,
- "Negative or too big volume value %f\n", d);
- return AVERROR(EINVAL);
- }
+ if ((ret = av_opt_set_from_string(vol, args, shorthand, "=", ":")) < 0)
+ return ret;
- vol->volume = d;
+ 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:(%d/256)(%f)(%1.2fdB) precision:fixed\n",
+ vol->volume_i, vol->volume, 20.0*log(vol->volume)/M_LN10);
+ } else {
+ av_log(ctx, AV_LOG_VERBOSE, "volume:(%f)(%1.2fdB) precision:%s\n",
+ vol->volume, 20.0*log(vol->volume)/M_LN10,
+ precision_str[vol->precision]);
}
- vol->volume_i = (int)(vol->volume * 256 + 0.5);
- av_log(ctx, AV_LOG_VERBOSE, "volume=%f\n", vol->volume);
- return 0;
+ av_opt_free(vol);
+ return ret;
}
static int query_formats(AVFilterContext *ctx)
{
+ VolumeContext *vol = ctx->priv;
AVFilterFormats *formats = NULL;
AVFilterChannelLayouts *layouts;
- enum AVSampleFormat sample_fmts[] = {
- AV_SAMPLE_FMT_U8,
- AV_SAMPLE_FMT_S16,
- AV_SAMPLE_FMT_S32,
- AV_SAMPLE_FMT_FLT,
- AV_SAMPLE_FMT_DBL,
- AV_SAMPLE_FMT_NONE
+ 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_layouts();
return AVERROR(ENOMEM);
ff_set_common_channel_layouts(ctx, layouts);
- formats = ff_make_format_list(sample_fmts);
+ formats = ff_make_format_list(sample_fmts[vol->precision]);
if (!formats)
return AVERROR(ENOMEM);
ff_set_common_formats(ctx, formats);
return 0;
}
-static int filter_frame(AVFilterLink *inlink, AVFilterBufferRef *insamples)
+static inline void scale_samples_u8(uint8_t *dst, const uint8_t *src,
+ int nb_samples, int volume)
{
- VolumeContext *vol = inlink->dst->priv;
- AVFilterLink *outlink = inlink->dst->outputs[0];
- const int nb_samples = insamples->audio->nb_samples *
- av_get_channel_layout_nb_channels(insamples->audio->channel_layout);
- const double volume = vol->volume;
- const int volume_i = vol->volume_i;
int i;
+ for (i = 0; i < nb_samples; i++)
+ dst[i] = av_clip_uint8(((((int64_t)src[i] - 128) * volume + 128) >> 8) + 128);
+}
- if (volume_i != 256) {
- switch (insamples->format) {
- case AV_SAMPLE_FMT_U8:
- {
- uint8_t *p = (void *)insamples->data[0];
- for (i = 0; i < nb_samples; i++) {
- int v = (((*p - 128) * volume_i + 128) >> 8) + 128;
- *p++ = av_clip_uint8(v);
- }
- break;
- }
- case AV_SAMPLE_FMT_S16:
- {
- int16_t *p = (void *)insamples->data[0];
- for (i = 0; i < nb_samples; i++) {
- int v = ((int64_t)*p * volume_i + 128) >> 8;
- *p++ = av_clip_int16(v);
- }
- break;
- }
- case AV_SAMPLE_FMT_S32:
- {
- int32_t *p = (void *)insamples->data[0];
- for (i = 0; i < nb_samples; i++) {
- int64_t v = (((int64_t)*p * volume_i + 128) >> 8);
- *p++ = av_clipl_int32(v);
+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 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:
+ avpriv_float_dsp_init(&vol->fdsp, 0);
+ vol->samples_align = 4;
+ break;
+ case AV_SAMPLE_FMT_DBL:
+ avpriv_float_dsp_init(&vol->fdsp, 0);
+ vol->samples_align = 8;
+ break;
+ }
+
+ if (ARCH_X86)
+ ff_volume_init_x86(vol);
+}
+
+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 = av_get_channel_layout_nb_channels(inlink->channel_layout);
+ vol->planes = av_sample_fmt_is_planar(inlink->format) ? vol->channels : 1;
+
+ volume_init(vol);
+
+ return 0;
+}
+
+static int filter_frame(AVFilterLink *inlink, AVFilterBufferRef *buf)
+{
+ VolumeContext *vol = inlink->dst->priv;
+ AVFilterLink *outlink = inlink->dst->outputs[0];
+ int nb_samples = buf->audio->nb_samples;
+ AVFilterBufferRef *out_buf;
+
+ if (vol->volume == 1.0 || vol->volume_i == 256)
+ return ff_filter_frame(outlink, buf);
+
+ /* do volume scaling in-place if input buffer is writable */
+ if (buf->perms & AV_PERM_WRITE) {
+ out_buf = buf;
+ } else {
+ out_buf = ff_get_audio_buffer(inlink, AV_PERM_WRITE, nb_samples);
+ if (!out_buf)
+ return AVERROR(ENOMEM);
+ out_buf->pts = buf->pts;
+ }
+
+ 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);
}
- break;
- }
- case AV_SAMPLE_FMT_FLT:
- {
- float *p = (void *)insamples->data[0];
- float scale = (float)volume;
- for (i = 0; i < nb_samples; i++) {
- *p++ *= scale;
+ } 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);
}
- break;
- }
- case AV_SAMPLE_FMT_DBL:
- {
- double *p = (void *)insamples->data[0];
- for (i = 0; i < nb_samples; i++) {
- *p *= volume;
- p++;
+ } 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);
}
- break;
- }
}
}
- return ff_filter_frame(outlink, insamples);
+
+ if (buf != out_buf)
+ avfilter_unref_buffer(buf);
+
+ return ff_filter_frame(outlink, out_buf);
}
-static const AVFilterPad volume_inputs[] = {
+static const AVFilterPad avfilter_af_volume_inputs[] = {
{
- .name = "default",
- .type = AVMEDIA_TYPE_AUDIO,
- .filter_frame = filter_frame,
- .min_perms = AV_PERM_READ | AV_PERM_WRITE,
+ .name = "default",
+ .type = AVMEDIA_TYPE_AUDIO,
+ .filter_frame = filter_frame,
},
- { NULL },
+ { NULL }
};
-static const AVFilterPad volume_outputs[] = {
+static const AVFilterPad avfilter_af_volume_outputs[] = {
{
- .name = "default",
- .type = AVMEDIA_TYPE_AUDIO,
+ .name = "default",
+ .type = AVMEDIA_TYPE_AUDIO,
+ .config_props = config_output,
},
- { NULL },
+ { NULL }
};
AVFilter avfilter_af_volume = {
.query_formats = query_formats,
.priv_size = sizeof(VolumeContext),
.init = init,
- .inputs = volume_inputs,
- .outputs = volume_outputs,
+ .inputs = avfilter_af_volume_inputs,
+ .outputs = avfilter_af_volume_outputs,
+ .priv_class = &volume_class,
};
projects / ffmpeg / blobdiff