mirror of https://github.com/FFmpeg/FFmpeg.git
Paul B Mahol
12 years ago
parent
4977e467a5
commit
884c890535
6 changed files with 423 additions and 2 deletions
@ -0,0 +1,357 @@ |
||||
/*
|
||||
* Copyright (c) 2013 Paul B Mahol |
||||
* |
||||
* 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 |
||||
* |
||||
*/ |
||||
|
||||
#include "libavutil/avstring.h" |
||||
#include "libavutil/opt.h" |
||||
#include "libavutil/samplefmt.h" |
||||
#include "libavutil/avassert.h" |
||||
#include "avfilter.h" |
||||
#include "audio.h" |
||||
#include "internal.h" |
||||
|
||||
typedef struct AudioEchoContext { |
||||
const AVClass *class; |
||||
float in_gain, out_gain; |
||||
char *delays, *decays; |
||||
float *delay, *decay; |
||||
int nb_echoes; |
||||
int delay_index; |
||||
uint8_t **delayptrs; |
||||
int max_samples, fade_out; |
||||
int *samples; |
||||
int64_t next_pts; |
||||
|
||||
void (*echo_samples)(struct AudioEchoContext *ctx, uint8_t **delayptrs, |
||||
uint8_t * const *src, uint8_t **dst, |
||||
int nb_samples, int channels); |
||||
} AudioEchoContext; |
||||
|
||||
#define OFFSET(x) offsetof(AudioEchoContext, x) |
||||
#define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM |
||||
|
||||
static const AVOption aecho_options[] = { |
||||
{ "in_gain", "set signal input gain", OFFSET(in_gain), AV_OPT_TYPE_FLOAT, {.dbl=0.6}, 0, 1, A }, |
||||
{ "out_gain", "set signal output gain", OFFSET(out_gain), AV_OPT_TYPE_FLOAT, {.dbl=0.3}, 0, 1, A }, |
||||
{ "delays", "set list of signal delays", OFFSET(delays), AV_OPT_TYPE_STRING, {.str="1000"}, 0, 0, A }, |
||||
{ "decays", "set list of signal decays", OFFSET(decays), AV_OPT_TYPE_STRING, {.str="0.5"}, 0, 0, A }, |
||||
{ NULL }, |
||||
}; |
||||
|
||||
AVFILTER_DEFINE_CLASS(aecho); |
||||
|
||||
static void count_items(char *item_str, int *nb_items) |
||||
{ |
||||
char *p; |
||||
|
||||
*nb_items = 1; |
||||
for (p = item_str; *p; p++) { |
||||
if (*p == '|') |
||||
(*nb_items)++; |
||||
} |
||||
|
||||
} |
||||
|
||||
static void fill_items(char *item_str, int *nb_items, float *items) |
||||
{ |
||||
char *p, *saveptr = NULL; |
||||
int i, new_nb_items = 0; |
||||
|
||||
p = item_str; |
||||
for (i = 0; i < *nb_items; i++) { |
||||
char *tstr = av_strtok(p, "|", &saveptr); |
||||
p = NULL; |
||||
new_nb_items += sscanf(tstr, "%f", &items[i]) == 1; |
||||
} |
||||
|
||||
*nb_items = new_nb_items; |
||||
} |
||||
|
||||
static av_cold void uninit(AVFilterContext *ctx) |
||||
{ |
||||
AudioEchoContext *s = ctx->priv; |
||||
|
||||
av_freep(&s->delay); |
||||
av_freep(&s->decay); |
||||
av_freep(&s->samples); |
||||
|
||||
if (s->delayptrs) |
||||
av_freep(s->delayptrs[0]); |
||||
av_freep(&s->delayptrs); |
||||
} |
||||
|
||||
static av_cold int init(AVFilterContext *ctx) |
||||
{ |
||||
AudioEchoContext *s = ctx->priv; |
||||
int nb_delays, nb_decays, i; |
||||
|
||||
if (!s->delays || !s->decays) { |
||||
av_log(ctx, AV_LOG_ERROR, "Missing delays and/or decays.\n"); |
||||
return AVERROR(EINVAL); |
||||
} |
||||
|
||||
count_items(s->delays, &nb_delays); |
||||
count_items(s->decays, &nb_decays); |
||||
|
||||
s->delay = av_realloc_f(s->delay, nb_delays, sizeof(*s->delay)); |
||||
s->decay = av_realloc_f(s->decay, nb_decays, sizeof(*s->decay)); |
||||
if (!s->delay || !s->decay) |
||||
return AVERROR(ENOMEM); |
||||
|
||||
fill_items(s->delays, &nb_delays, s->delay); |
||||
fill_items(s->decays, &nb_decays, s->decay); |
||||
|
||||
if (nb_delays != nb_decays) { |
||||
av_log(ctx, AV_LOG_ERROR, "Number of delays %d differs from number of decays %d.\n", nb_delays, nb_decays); |
||||
return AVERROR(EINVAL); |
||||
} |
||||
|
||||
s->nb_echoes = nb_delays; |
||||
if (!s->nb_echoes) { |
||||
av_log(ctx, AV_LOG_ERROR, "At least one decay & delay must be set.\n"); |
||||
return AVERROR(EINVAL); |
||||
} |
||||
|
||||
s->samples = av_realloc_f(s->samples, nb_delays, sizeof(*s->samples)); |
||||
if (!s->samples) |
||||
return AVERROR(ENOMEM); |
||||
|
||||
for (i = 0; i < nb_delays; i++) { |
||||
if (s->delay[i] <= 0 || s->delay[i] > 90000) { |
||||
av_log(ctx, AV_LOG_ERROR, "delay[%d]: %f is out of allowed range: (0, 90000]\n", i, s->delay[i]); |
||||
return AVERROR(EINVAL); |
||||
} |
||||
if (s->decay[i] <= 0 || s->decay[i] > 1) { |
||||
av_log(ctx, AV_LOG_ERROR, "decay[%d]: %f is out of allowed range: (0, 1]\n", i, s->decay[i]); |
||||
return AVERROR(EINVAL); |
||||
} |
||||
} |
||||
|
||||
s->next_pts = AV_NOPTS_VALUE; |
||||
|
||||
av_log(ctx, AV_LOG_DEBUG, "nb_echoes:%d\n", s->nb_echoes); |
||||
return 0; |
||||
} |
||||
|
||||
static int query_formats(AVFilterContext *ctx) |
||||
{ |
||||
AVFilterChannelLayouts *layouts; |
||||
AVFilterFormats *formats; |
||||
static const enum AVSampleFormat sample_fmts[] = { |
||||
AV_SAMPLE_FMT_S16P, AV_SAMPLE_FMT_S32P, |
||||
AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_DBLP, |
||||
AV_SAMPLE_FMT_NONE |
||||
}; |
||||
|
||||
layouts = ff_all_channel_layouts(); |
||||
if (!layouts) |
||||
return AVERROR(ENOMEM); |
||||
ff_set_common_channel_layouts(ctx, layouts); |
||||
|
||||
formats = ff_make_format_list(sample_fmts); |
||||
if (!formats) |
||||
return AVERROR(ENOMEM); |
||||
ff_set_common_formats(ctx, formats); |
||||
|
||||
formats = ff_all_samplerates(); |
||||
if (!formats) |
||||
return AVERROR(ENOMEM); |
||||
ff_set_common_samplerates(ctx, formats); |
||||
|
||||
return 0; |
||||
} |
||||
|
||||
#define MOD(a, b) (((a) >= (b)) ? (a) - (b) : (a)) |
||||
|
||||
#define ECHO(name, type, min, max) \ |
||||
static void echo_samples_## name ##p(AudioEchoContext *ctx, \
|
||||
uint8_t **delayptrs, \
|
||||
uint8_t * const *src, uint8_t **dst, \
|
||||
int nb_samples, int channels) \
|
||||
{ \
|
||||
const double out_gain = ctx->out_gain; \
|
||||
const double in_gain = ctx->in_gain; \
|
||||
const int nb_echoes = ctx->nb_echoes; \
|
||||
const int max_samples = ctx->max_samples; \
|
||||
int i, j, chan, index; \
|
||||
\
|
||||
for (chan = 0; chan < channels; chan++) { \
|
||||
const type *s = (type *)src[chan]; \
|
||||
type *d = (type *)dst[chan]; \
|
||||
type *dbuf = (type *)delayptrs[chan]; \
|
||||
\
|
||||
index = ctx->delay_index; \
|
||||
for (i = 0; i < nb_samples; i++, s++, d++) { \
|
||||
double out, in; \
|
||||
\
|
||||
in = *s; \
|
||||
out = in * in_gain; \
|
||||
for (j = 0; j < nb_echoes; j++) { \
|
||||
int ix = index + max_samples - ctx->samples[j]; \
|
||||
ix = MOD(ix, max_samples); \
|
||||
out += dbuf[ix] * ctx->decay[j]; \
|
||||
} \
|
||||
out *= out_gain; \
|
||||
\
|
||||
*d = av_clipd(out, min, max); \
|
||||
dbuf[index] = in; \
|
||||
\
|
||||
index = MOD(index + 1, max_samples); \
|
||||
} \
|
||||
} \
|
||||
ctx->delay_index = index; \
|
||||
} |
||||
|
||||
ECHO(dbl, double, -1.0, 1.0 ) |
||||
ECHO(flt, float, -1.0, 1.0 ) |
||||
ECHO(s16, int16_t, INT16_MIN, INT16_MAX) |
||||
ECHO(s32, int32_t, INT32_MIN, INT32_MAX) |
||||
|
||||
static int config_output(AVFilterLink *outlink) |
||||
{ |
||||
AVFilterContext *ctx = outlink->src; |
||||
AudioEchoContext *s = ctx->priv; |
||||
float volume = 1.0; |
||||
int i; |
||||
|
||||
for (i = 0; i < s->nb_echoes; i++) { |
||||
s->samples[i] = s->delay[i] * outlink->sample_rate / 1000.0; |
||||
s->max_samples = FFMAX(s->max_samples, s->samples[i]); |
||||
volume += s->decay[i]; |
||||
} |
||||
|
||||
if (s->max_samples <= 0) { |
||||
av_log(ctx, AV_LOG_ERROR, "Nothing to echo - missing delay samples.\n"); |
||||
return AVERROR(EINVAL); |
||||
} |
||||
s->fade_out = s->max_samples; |
||||
|
||||
if (volume * s->in_gain * s->out_gain > 1.0) |
||||
av_log(ctx, AV_LOG_WARNING, |
||||
"out_gain %f can cause saturation of output\n", s->out_gain); |
||||
|
||||
switch (outlink->format) { |
||||
case AV_SAMPLE_FMT_DBLP: s->echo_samples = echo_samples_dblp; break; |
||||
case AV_SAMPLE_FMT_FLTP: s->echo_samples = echo_samples_fltp; break; |
||||
case AV_SAMPLE_FMT_S16P: s->echo_samples = echo_samples_s16p; break; |
||||
case AV_SAMPLE_FMT_S32P: s->echo_samples = echo_samples_s32p; break; |
||||
} |
||||
|
||||
|
||||
if (s->delayptrs) |
||||
av_freep(s->delayptrs[0]); |
||||
av_freep(&s->delayptrs); |
||||
|
||||
return av_samples_alloc_array_and_samples(&s->delayptrs, NULL, |
||||
outlink->channels, |
||||
s->max_samples, |
||||
outlink->format, 0); |
||||
} |
||||
|
||||
static int filter_frame(AVFilterLink *inlink, AVFrame *frame) |
||||
{ |
||||
AVFilterContext *ctx = inlink->dst; |
||||
AudioEchoContext *s = ctx->priv; |
||||
AVFrame *out_frame; |
||||
|
||||
if (av_frame_is_writable(frame)) { |
||||
out_frame = frame; |
||||
} else { |
||||
out_frame = ff_get_audio_buffer(inlink, frame->nb_samples); |
||||
if (!out_frame) |
||||
return AVERROR(ENOMEM); |
||||
av_frame_copy_props(out_frame, frame); |
||||
} |
||||
|
||||
s->echo_samples(s, s->delayptrs, frame->data, out_frame->data, |
||||
frame->nb_samples, inlink->channels); |
||||
|
||||
if (frame != out_frame) |
||||
av_frame_free(&frame); |
||||
|
||||
s->next_pts = frame->pts + av_rescale_q(frame->nb_samples, (AVRational){1, inlink->sample_rate}, inlink->time_base); |
||||
return ff_filter_frame(ctx->outputs[0], out_frame); |
||||
} |
||||
|
||||
static int request_frame(AVFilterLink *outlink) |
||||
{ |
||||
AVFilterContext *ctx = outlink->src; |
||||
AudioEchoContext *s = ctx->priv; |
||||
int ret; |
||||
|
||||
ret = ff_request_frame(ctx->inputs[0]); |
||||
|
||||
if (ret == AVERROR_EOF && !ctx->is_disabled && s->fade_out) { |
||||
int nb_samples = FFMIN(s->fade_out, 2048); |
||||
AVFrame *frame; |
||||
|
||||
frame = ff_get_audio_buffer(outlink, nb_samples); |
||||
if (!frame) |
||||
return AVERROR(ENOMEM); |
||||
s->fade_out -= nb_samples; |
||||
|
||||
av_samples_set_silence(frame->extended_data, 0, |
||||
frame->nb_samples, |
||||
outlink->channels, |
||||
frame->format); |
||||
|
||||
s->echo_samples(s, s->delayptrs, frame->data, frame->data, |
||||
frame->nb_samples, outlink->channels); |
||||
|
||||
frame->pts = s->next_pts; |
||||
if (s->next_pts != AV_NOPTS_VALUE) |
||||
s->next_pts += av_rescale_q(nb_samples, (AVRational){1, outlink->sample_rate}, outlink->time_base); |
||||
|
||||
return ff_filter_frame(outlink, frame); |
||||
} |
||||
|
||||
return ret; |
||||
} |
||||
|
||||
static const AVFilterPad aecho_inputs[] = { |
||||
{ |
||||
.name = "default", |
||||
.type = AVMEDIA_TYPE_AUDIO, |
||||
.filter_frame = filter_frame, |
||||
}, |
||||
{ NULL }, |
||||
}; |
||||
|
||||
static const AVFilterPad aecho_outputs[] = { |
||||
{ |
||||
.name = "default", |
||||
.request_frame = request_frame, |
||||
.config_props = config_output, |
||||
.type = AVMEDIA_TYPE_AUDIO, |
||||
}, |
||||
{ NULL }, |
||||
}; |
||||
|
||||
AVFilter avfilter_af_aecho = { |
||||
.name = "aecho", |
||||
.description = NULL_IF_CONFIG_SMALL("Add echoing to the audio."), |
||||
.query_formats = query_formats, |
||||
.priv_size = sizeof(AudioEchoContext), |
||||
.priv_class = &aecho_class, |
||||
.init = init, |
||||
.uninit = uninit, |
||||
.inputs = aecho_inputs, |
||||
.outputs = aecho_outputs, |
||||
}; |
||||
Loading…
Reference in new issue