From 9d05de2258769993c289395d3f8bf41b7a3138af Mon Sep 17 00:00:00 2001 From: Paul B Mahol Date: Fri, 13 Sep 2013 11:36:52 +0000 Subject: avfilter: add adelay filter Signed-off-by: Paul B Mahol --- libavfilter/af_adelay.c | 283 ++++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 283 insertions(+) create mode 100644 libavfilter/af_adelay.c (limited to 'libavfilter/af_adelay.c') diff --git a/libavfilter/af_adelay.c b/libavfilter/af_adelay.c new file mode 100644 index 0000000000..d51264fb46 --- /dev/null +++ b/libavfilter/af_adelay.c @@ -0,0 +1,283 @@ +/* + * 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 "avfilter.h" +#include "audio.h" +#include "internal.h" + +typedef struct ChanDelay { + int delay; + unsigned delay_index; + unsigned index; + uint8_t *samples; +} ChanDelay; + +typedef struct AudioDelayContext { + const AVClass *class; + char *delays; + ChanDelay *chandelay; + int nb_delays; + int block_align; + unsigned max_delay; + int64_t next_pts; + + void (*delay_channel)(ChanDelay *d, int nb_samples, + const uint8_t *src, uint8_t *dst); +} AudioDelayContext; + +#define OFFSET(x) offsetof(AudioDelayContext, x) +#define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM + +static const AVOption adelay_options[] = { + { "delays", "set list of delays for each channel", OFFSET(delays), AV_OPT_TYPE_STRING, {.str=NULL}, 0, 0, A }, + { NULL } +}; + +AVFILTER_DEFINE_CLASS(adelay); + +static int query_formats(AVFilterContext *ctx) +{ + AVFilterChannelLayouts *layouts; + AVFilterFormats *formats; + static const enum AVSampleFormat sample_fmts[] = { + AV_SAMPLE_FMT_U8P, 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 DELAY(name, type, fill) \ +static void delay_channel_## name ##p(ChanDelay *d, int nb_samples, \ + const uint8_t *ssrc, uint8_t *ddst) \ +{ \ + const type *src = (type *)ssrc; \ + type *dst = (type *)ddst; \ + type *samples = (type *)d->samples; \ + \ + while (nb_samples) { \ + if (d->delay_index < d->delay) { \ + const int len = FFMIN(nb_samples, d->delay - d->delay_index); \ + \ + memcpy(&samples[d->delay_index], src, len * sizeof(type)); \ + memset(dst, fill, len * sizeof(type)); \ + d->delay_index += len; \ + src += len; \ + dst += len; \ + nb_samples -= len; \ + } else { \ + *dst = samples[d->index]; \ + samples[d->index] = *src; \ + nb_samples--; \ + d->index++; \ + src++, dst++; \ + d->index = d->index >= d->delay ? 0 : d->index; \ + } \ + } \ +} + +DELAY(u8, uint8_t, 0x80) +DELAY(s16, int16_t, 0) +DELAY(s32, int32_t, 0) +DELAY(flt, float, 0) +DELAY(dbl, double, 0) + +static int config_input(AVFilterLink *inlink) +{ + AVFilterContext *ctx = inlink->dst; + AudioDelayContext *s = ctx->priv; + char *p, *arg, *saveptr = NULL; + int i; + + s->chandelay = av_calloc(inlink->channels, sizeof(*s->chandelay)); + if (!s->chandelay) + return AVERROR(ENOMEM); + s->nb_delays = inlink->channels; + s->block_align = av_get_bytes_per_sample(inlink->format); + + p = s->delays; + for (i = 0; i < s->nb_delays; i++) { + ChanDelay *d = &s->chandelay[i]; + float delay; + + if (!(arg = av_strtok(p, "|", &saveptr))) + break; + + p = NULL; + sscanf(arg, "%f", &delay); + + d->delay = delay * inlink->sample_rate / 1000.0; + if (d->delay < 0) { + av_log(ctx, AV_LOG_ERROR, "Delay must be non negative number.\n"); + return AVERROR(EINVAL); + } + } + + for (i = 0; i < s->nb_delays; i++) { + ChanDelay *d = &s->chandelay[i]; + + if (!d->delay) + continue; + + d->samples = av_malloc_array(d->delay, s->block_align); + if (!d->samples) + return AVERROR(ENOMEM); + + s->max_delay = FFMAX(s->max_delay, d->delay); + } + + if (!s->max_delay) { + av_log(ctx, AV_LOG_ERROR, "At least one delay >0 must be specified.\n"); + return AVERROR(EINVAL); + } + + switch (inlink->format) { + case AV_SAMPLE_FMT_U8P : s->delay_channel = delay_channel_u8p ; break; + case AV_SAMPLE_FMT_S16P: s->delay_channel = delay_channel_s16p; break; + case AV_SAMPLE_FMT_S32P: s->delay_channel = delay_channel_s32p; break; + case AV_SAMPLE_FMT_FLTP: s->delay_channel = delay_channel_fltp; break; + case AV_SAMPLE_FMT_DBLP: s->delay_channel = delay_channel_dblp; break; + } + + return 0; +} + +static int filter_frame(AVFilterLink *inlink, AVFrame *frame) +{ + AVFilterContext *ctx = inlink->dst; + AudioDelayContext *s = ctx->priv; + AVFrame *out_frame; + int i; + + if (ctx->is_disabled || !s->delays) + return ff_filter_frame(ctx->outputs[0], frame); + + out_frame = ff_get_audio_buffer(inlink, frame->nb_samples); + if (!out_frame) + return AVERROR(ENOMEM); + av_frame_copy_props(out_frame, frame); + + for (i = 0; i < s->nb_delays; i++) { + ChanDelay *d = &s->chandelay[i]; + const uint8_t *src = frame->extended_data[i]; + uint8_t *dst = out_frame->extended_data[i]; + + if (!d->delay) + memcpy(dst, src, frame->nb_samples * s->block_align); + else + s->delay_channel(d, frame->nb_samples, src, dst); + } + + s->next_pts = frame->pts + av_rescale_q(frame->nb_samples, (AVRational){1, inlink->sample_rate}, inlink->time_base); + av_frame_free(&frame); + return ff_filter_frame(ctx->outputs[0], out_frame); +} + +static int request_frame(AVFilterLink *outlink) +{ + AVFilterContext *ctx = outlink->src; + AudioDelayContext *s = ctx->priv; + int ret; + + ret = ff_request_frame(ctx->inputs[0]); + if (ret == AVERROR_EOF && !ctx->is_disabled && s->max_delay) { + int nb_samples = FFMIN(s->max_delay, 2048); + AVFrame *frame; + + frame = ff_get_audio_buffer(outlink, nb_samples); + if (!frame) + return AVERROR(ENOMEM); + s->max_delay -= nb_samples; + + av_samples_set_silence(frame->extended_data, 0, + frame->nb_samples, + outlink->channels, + frame->format); + + 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); + + ret = filter_frame(ctx->inputs[0], frame); + } + + return ret; +} + +static av_cold void uninit(AVFilterContext *ctx) +{ + AudioDelayContext *s = ctx->priv; + int i; + + for (i = 0; i < s->nb_delays; i++) + av_free(s->chandelay[i].samples); + av_freep(&s->chandelay); +} + +static const AVFilterPad adelay_inputs[] = { + { + .name = "default", + .type = AVMEDIA_TYPE_AUDIO, + .config_props = config_input, + .filter_frame = filter_frame, + }, + { NULL } +}; + +static const AVFilterPad adelay_outputs[] = { + { + .name = "default", + .request_frame = request_frame, + .type = AVMEDIA_TYPE_AUDIO, + }, + { NULL } +}; + +AVFilter avfilter_af_adelay = { + .name = "adelay", + .description = NULL_IF_CONFIG_SMALL("Delay one or more audio channels."), + .query_formats = query_formats, + .priv_size = sizeof(AudioDelayContext), + .priv_class = &adelay_class, + .uninit = uninit, + .inputs = adelay_inputs, + .outputs = adelay_outputs, + .flags = AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL, +}; -- cgit v1.2.3