/* * This file is part of Libav. * * Libav 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. * * Libav 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 Libav; if not, write to the Free Software * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ #include "libavresample/avresample.h" #include "libavutil/audio_fifo.h" #include "libavutil/common.h" #include "libavutil/mathematics.h" #include "libavutil/opt.h" #include "libavutil/samplefmt.h" #include "audio.h" #include "avfilter.h" #include "internal.h" typedef struct ASyncContext { const AVClass *class; AVAudioResampleContext *avr; int64_t pts; ///< timestamp in samples of the first sample in fifo int min_delta; ///< pad/trim min threshold in samples /* options */ int resample; float min_delta_sec; int max_comp; /* set by filter_samples() to signal an output frame to request_frame() */ int got_output; } ASyncContext; #define OFFSET(x) offsetof(ASyncContext, x) #define A AV_OPT_FLAG_AUDIO_PARAM static const AVOption options[] = { { "compensate", "Stretch/squeeze the data to make it match the timestamps", OFFSET(resample), AV_OPT_TYPE_INT, { 0 }, 0, 1, A }, { "min_delta", "Minimum difference between timestamps and audio data " "(in seconds) to trigger padding/trimmin the data.", OFFSET(min_delta_sec), AV_OPT_TYPE_FLOAT, { 0.1 }, 0, INT_MAX, A }, { "max_comp", "Maximum compensation in samples per second.", OFFSET(max_comp), AV_OPT_TYPE_INT, { 500 }, 0, INT_MAX, A }, { "first_pts", "Assume the first pts should be this value.", OFFSET(pts), AV_OPT_TYPE_INT64, { AV_NOPTS_VALUE }, INT64_MIN, INT64_MAX, A }, { NULL }, }; static const AVClass async_class = { .class_name = "asyncts filter", .item_name = av_default_item_name, .option = options, .version = LIBAVUTIL_VERSION_INT, }; static int init(AVFilterContext *ctx, const char *args) { ASyncContext *s = ctx->priv; int ret; s->class = &async_class; av_opt_set_defaults(s); if ((ret = av_set_options_string(s, args, "=", ":")) < 0) { av_log(ctx, AV_LOG_ERROR, "Error parsing options string '%s'.\n", args); return ret; } av_opt_free(s); return 0; } static void uninit(AVFilterContext *ctx) { ASyncContext *s = ctx->priv; if (s->avr) { avresample_close(s->avr); avresample_free(&s->avr); } } static int config_props(AVFilterLink *link) { ASyncContext *s = link->src->priv; int ret; s->min_delta = s->min_delta_sec * link->sample_rate; link->time_base = (AVRational){1, link->sample_rate}; s->avr = avresample_alloc_context(); if (!s->avr) return AVERROR(ENOMEM); av_opt_set_int(s->avr, "in_channel_layout", link->channel_layout, 0); av_opt_set_int(s->avr, "out_channel_layout", link->channel_layout, 0); av_opt_set_int(s->avr, "in_sample_fmt", link->format, 0); av_opt_set_int(s->avr, "out_sample_fmt", link->format, 0); av_opt_set_int(s->avr, "in_sample_rate", link->sample_rate, 0); av_opt_set_int(s->avr, "out_sample_rate", link->sample_rate, 0); if (s->resample) av_opt_set_int(s->avr, "force_resampling", 1, 0); if ((ret = avresample_open(s->avr)) < 0) return ret; return 0; } static int request_frame(AVFilterLink *link) { AVFilterContext *ctx = link->src; ASyncContext *s = ctx->priv; int ret = 0; int nb_samples; s->got_output = 0; while (ret >= 0 && !s->got_output) ret = ff_request_frame(ctx->inputs[0]); /* flush the fifo */ if (ret == AVERROR_EOF && (nb_samples = avresample_get_delay(s->avr))) { AVFilterBufferRef *buf = ff_get_audio_buffer(link, AV_PERM_WRITE, nb_samples); if (!buf) return AVERROR(ENOMEM); avresample_convert(s->avr, (void**)buf->extended_data, buf->linesize[0], nb_samples, NULL, 0, 0); buf->pts = s->pts; return ff_filter_samples(link, buf); } return ret; } static int write_to_fifo(ASyncContext *s, AVFilterBufferRef *buf) { int ret = avresample_convert(s->avr, NULL, 0, 0, (void**)buf->extended_data, buf->linesize[0], buf->audio->nb_samples); avfilter_unref_buffer(buf); return ret; } /* get amount of data currently buffered, in samples */ static int64_t get_delay(ASyncContext *s) { return avresample_available(s->avr) + avresample_get_delay(s->avr); } static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf) { AVFilterContext *ctx = inlink->dst; ASyncContext *s = ctx->priv; AVFilterLink *outlink = ctx->outputs[0]; int nb_channels = av_get_channel_layout_nb_channels(buf->audio->channel_layout); int64_t pts = (buf->pts == AV_NOPTS_VALUE) ? buf->pts : av_rescale_q(buf->pts, inlink->time_base, outlink->time_base); int out_size, ret; int64_t delta; /* buffer data until we get the first timestamp */ if (s->pts == AV_NOPTS_VALUE) { if (pts != AV_NOPTS_VALUE) { s->pts = pts - get_delay(s); } return write_to_fifo(s, buf); } /* now wait for the next timestamp */ if (pts == AV_NOPTS_VALUE) { return write_to_fifo(s, buf); } /* when we have two timestamps, compute how many samples would we have * to add/remove to get proper sync between data and timestamps */ delta = pts - s->pts - get_delay(s); out_size = avresample_available(s->avr); if (labs(delta) > s->min_delta) { av_log(ctx, AV_LOG_VERBOSE, "Discontinuity - %"PRId64" samples.\n", delta); out_size = av_clipl_int32((int64_t)out_size + delta); } else { if (s->resample) { int comp = av_clip(delta, -s->max_comp, s->max_comp); av_log(ctx, AV_LOG_VERBOSE, "Compensating %d samples per second.\n", comp); avresample_set_compensation(s->avr, delta, inlink->sample_rate); } delta = 0; } if (out_size > 0) { AVFilterBufferRef *buf_out = ff_get_audio_buffer(outlink, AV_PERM_WRITE, out_size); if (!buf_out) { ret = AVERROR(ENOMEM); goto fail; } avresample_read(s->avr, (void**)buf_out->extended_data, out_size); buf_out->pts = s->pts; if (delta > 0) { av_samples_set_silence(buf_out->extended_data, out_size - delta, delta, nb_channels, buf->format); } ret = ff_filter_samples(outlink, buf_out); if (ret < 0) goto fail; s->got_output = 1; } else { av_log(ctx, AV_LOG_WARNING, "Non-monotonous timestamps, dropping " "whole buffer.\n"); } /* drain any remaining buffered data */ avresample_read(s->avr, NULL, avresample_available(s->avr)); s->pts = pts - avresample_get_delay(s->avr); ret = avresample_convert(s->avr, NULL, 0, 0, (void**)buf->extended_data, buf->linesize[0], buf->audio->nb_samples); fail: avfilter_unref_buffer(buf); return ret; } AVFilter avfilter_af_asyncts = { .name = "asyncts", .description = NULL_IF_CONFIG_SMALL("Sync audio data to timestamps"), .init = init, .uninit = uninit, .priv_size = sizeof(ASyncContext), .inputs = (const AVFilterPad[]) {{ .name = "default", .type = AVMEDIA_TYPE_AUDIO, .filter_samples = filter_samples }, { NULL }}, .outputs = (const AVFilterPad[]) {{ .name = "default", .type = AVMEDIA_TYPE_AUDIO, .config_props = config_props, .request_frame = request_frame }, { NULL }}, };