mirror of https://github.com/FFmpeg/FFmpeg.git
Paul B Mahol
11 years ago
parent
42b8f5fba1
commit
9d05de2258
6 changed files with 315 additions and 1 deletions
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/*
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* Copyright (c) 2013 Paul B Mahol |
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* |
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* This file is part of FFmpeg. |
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* |
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* FFmpeg is free software; you can redistribute it and/or |
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* modify it under the terms of the GNU Lesser General Public |
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* License as published by the Free Software Foundation; either |
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* version 2.1 of the License, or (at your option) any later version. |
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* |
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* FFmpeg is distributed in the hope that it will be useful, |
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* but WITHOUT ANY WARRANTY; without even the implied warranty of |
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU |
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* Lesser General Public License for more details. |
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* |
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* You should have received a copy of the GNU Lesser General Public |
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* License along with FFmpeg; if not, write to the Free Software |
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA |
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* |
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*/ |
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#include "libavutil/avstring.h" |
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#include "libavutil/opt.h" |
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#include "libavutil/samplefmt.h" |
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#include "avfilter.h" |
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#include "audio.h" |
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#include "internal.h" |
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typedef struct ChanDelay { |
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int delay; |
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unsigned delay_index; |
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unsigned index; |
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uint8_t *samples; |
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} ChanDelay; |
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typedef struct AudioDelayContext { |
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const AVClass *class; |
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char *delays; |
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ChanDelay *chandelay; |
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int nb_delays; |
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int block_align; |
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unsigned max_delay; |
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int64_t next_pts; |
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void (*delay_channel)(ChanDelay *d, int nb_samples, |
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const uint8_t *src, uint8_t *dst); |
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} AudioDelayContext; |
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#define OFFSET(x) offsetof(AudioDelayContext, x) |
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#define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM |
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static const AVOption adelay_options[] = { |
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{ "delays", "set list of delays for each channel", OFFSET(delays), AV_OPT_TYPE_STRING, {.str=NULL}, 0, 0, A }, |
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{ NULL } |
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}; |
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AVFILTER_DEFINE_CLASS(adelay); |
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static int query_formats(AVFilterContext *ctx) |
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{ |
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AVFilterChannelLayouts *layouts; |
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AVFilterFormats *formats; |
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static const enum AVSampleFormat sample_fmts[] = { |
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AV_SAMPLE_FMT_U8P, AV_SAMPLE_FMT_S16P, AV_SAMPLE_FMT_S32P, |
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AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_DBLP, |
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AV_SAMPLE_FMT_NONE |
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}; |
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layouts = ff_all_channel_layouts(); |
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if (!layouts) |
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return AVERROR(ENOMEM); |
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ff_set_common_channel_layouts(ctx, layouts); |
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formats = ff_make_format_list(sample_fmts); |
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if (!formats) |
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return AVERROR(ENOMEM); |
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ff_set_common_formats(ctx, formats); |
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formats = ff_all_samplerates(); |
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if (!formats) |
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return AVERROR(ENOMEM); |
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ff_set_common_samplerates(ctx, formats); |
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return 0; |
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} |
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#define DELAY(name, type, fill) \ |
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static void delay_channel_## name ##p(ChanDelay *d, int nb_samples, \
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const uint8_t *ssrc, uint8_t *ddst) \
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{ \
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const type *src = (type *)ssrc; \
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type *dst = (type *)ddst; \
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type *samples = (type *)d->samples; \
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\
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while (nb_samples) { \
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if (d->delay_index < d->delay) { \
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const int len = FFMIN(nb_samples, d->delay - d->delay_index); \
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\
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memcpy(&samples[d->delay_index], src, len * sizeof(type)); \
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memset(dst, fill, len * sizeof(type)); \
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d->delay_index += len; \
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src += len; \
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dst += len; \
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nb_samples -= len; \
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} else { \
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*dst = samples[d->index]; \
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samples[d->index] = *src; \
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nb_samples--; \
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d->index++; \
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src++, dst++; \
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d->index = d->index >= d->delay ? 0 : d->index; \
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} \
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} \
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} |
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DELAY(u8, uint8_t, 0x80) |
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DELAY(s16, int16_t, 0) |
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DELAY(s32, int32_t, 0) |
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DELAY(flt, float, 0) |
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DELAY(dbl, double, 0) |
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static int config_input(AVFilterLink *inlink) |
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{ |
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AVFilterContext *ctx = inlink->dst; |
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AudioDelayContext *s = ctx->priv; |
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char *p, *arg, *saveptr = NULL; |
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int i; |
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s->chandelay = av_calloc(inlink->channels, sizeof(*s->chandelay)); |
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if (!s->chandelay) |
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return AVERROR(ENOMEM); |
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s->nb_delays = inlink->channels; |
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s->block_align = av_get_bytes_per_sample(inlink->format); |
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p = s->delays; |
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for (i = 0; i < s->nb_delays; i++) { |
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ChanDelay *d = &s->chandelay[i]; |
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float delay; |
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if (!(arg = av_strtok(p, "|", &saveptr))) |
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break; |
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p = NULL; |
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sscanf(arg, "%f", &delay); |
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d->delay = delay * inlink->sample_rate / 1000.0; |
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if (d->delay < 0) { |
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av_log(ctx, AV_LOG_ERROR, "Delay must be non negative number.\n"); |
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return AVERROR(EINVAL); |
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} |
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} |
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for (i = 0; i < s->nb_delays; i++) { |
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ChanDelay *d = &s->chandelay[i]; |
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if (!d->delay) |
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continue; |
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d->samples = av_malloc_array(d->delay, s->block_align); |
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if (!d->samples) |
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return AVERROR(ENOMEM); |
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s->max_delay = FFMAX(s->max_delay, d->delay); |
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} |
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if (!s->max_delay) { |
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av_log(ctx, AV_LOG_ERROR, "At least one delay >0 must be specified.\n"); |
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return AVERROR(EINVAL); |
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} |
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switch (inlink->format) { |
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case AV_SAMPLE_FMT_U8P : s->delay_channel = delay_channel_u8p ; break; |
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case AV_SAMPLE_FMT_S16P: s->delay_channel = delay_channel_s16p; break; |
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case AV_SAMPLE_FMT_S32P: s->delay_channel = delay_channel_s32p; break; |
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case AV_SAMPLE_FMT_FLTP: s->delay_channel = delay_channel_fltp; break; |
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case AV_SAMPLE_FMT_DBLP: s->delay_channel = delay_channel_dblp; break; |
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} |
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return 0; |
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} |
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static int filter_frame(AVFilterLink *inlink, AVFrame *frame) |
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{ |
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AVFilterContext *ctx = inlink->dst; |
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AudioDelayContext *s = ctx->priv; |
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AVFrame *out_frame; |
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int i; |
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if (ctx->is_disabled || !s->delays) |
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return ff_filter_frame(ctx->outputs[0], frame); |
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out_frame = ff_get_audio_buffer(inlink, frame->nb_samples); |
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if (!out_frame) |
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return AVERROR(ENOMEM); |
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av_frame_copy_props(out_frame, frame); |
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for (i = 0; i < s->nb_delays; i++) { |
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ChanDelay *d = &s->chandelay[i]; |
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const uint8_t *src = frame->extended_data[i]; |
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uint8_t *dst = out_frame->extended_data[i]; |
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if (!d->delay) |
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memcpy(dst, src, frame->nb_samples * s->block_align); |
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else |
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s->delay_channel(d, frame->nb_samples, src, dst); |
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} |
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s->next_pts = frame->pts + av_rescale_q(frame->nb_samples, (AVRational){1, inlink->sample_rate}, inlink->time_base); |
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av_frame_free(&frame); |
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return ff_filter_frame(ctx->outputs[0], out_frame); |
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} |
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static int request_frame(AVFilterLink *outlink) |
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{ |
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AVFilterContext *ctx = outlink->src; |
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AudioDelayContext *s = ctx->priv; |
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int ret; |
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ret = ff_request_frame(ctx->inputs[0]); |
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if (ret == AVERROR_EOF && !ctx->is_disabled && s->max_delay) { |
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int nb_samples = FFMIN(s->max_delay, 2048); |
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AVFrame *frame; |
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frame = ff_get_audio_buffer(outlink, nb_samples); |
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if (!frame) |
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return AVERROR(ENOMEM); |
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s->max_delay -= nb_samples; |
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av_samples_set_silence(frame->extended_data, 0, |
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frame->nb_samples, |
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outlink->channels, |
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frame->format); |
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frame->pts = s->next_pts; |
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if (s->next_pts != AV_NOPTS_VALUE) |
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s->next_pts += av_rescale_q(nb_samples, (AVRational){1, outlink->sample_rate}, outlink->time_base); |
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ret = filter_frame(ctx->inputs[0], frame); |
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} |
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return ret; |
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} |
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static av_cold void uninit(AVFilterContext *ctx) |
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{ |
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AudioDelayContext *s = ctx->priv; |
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int i; |
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for (i = 0; i < s->nb_delays; i++) |
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av_free(s->chandelay[i].samples); |
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av_freep(&s->chandelay); |
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} |
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static const AVFilterPad adelay_inputs[] = { |
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{ |
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.name = "default", |
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.type = AVMEDIA_TYPE_AUDIO, |
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.config_props = config_input, |
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.filter_frame = filter_frame, |
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}, |
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{ NULL } |
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}; |
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static const AVFilterPad adelay_outputs[] = { |
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{ |
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.name = "default", |
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.request_frame = request_frame, |
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.type = AVMEDIA_TYPE_AUDIO, |
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}, |
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{ NULL } |
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}; |
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AVFilter avfilter_af_adelay = { |
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.name = "adelay", |
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.description = NULL_IF_CONFIG_SMALL("Delay one or more audio channels."), |
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.query_formats = query_formats, |
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.priv_size = sizeof(AudioDelayContext), |
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.priv_class = &adelay_class, |
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.uninit = uninit, |
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.inputs = adelay_inputs, |
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.outputs = adelay_outputs, |
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.flags = AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL, |
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}; |
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