diff options
Diffstat (limited to 'libavfilter/af_aecho.c')
| -rw-r--r-- | libavfilter/af_aecho.c | 359 |
1 files changed, 359 insertions, 0 deletions
diff --git a/libavfilter/af_aecho.c b/libavfilter/af_aecho.c new file mode 100644 index 0000000..c26fdd4 --- /dev/null +++ b/libavfilter/af_aecho.c @@ -0,0 +1,359 @@ +/* + * 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/avassert.h" +#include "libavutil/avstring.h" +#include "libavutil/opt.h" +#include "libavutil/samplefmt.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, av_uninit(index); \ + \ + av_assert1(channels > 0); /* would corrupt delay_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->extended_data, out_frame->extended_data, + frame->nb_samples, inlink->channels); + + s->next_pts = frame->pts + av_rescale_q(frame->nb_samples, (AVRational){1, inlink->sample_rate}, inlink->time_base); + + if (frame != out_frame) + av_frame_free(&frame); + + 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->extended_data, frame->extended_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 ff_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, +}; |
