/* * 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 <stdint.h> #include "libavresample/avresample.h" #include "libavutil/attributes.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 int first_frame; ///< 1 until filter_frame() has processed at least 1 frame with a pts != AV_NOPTS_VALUE int64_t first_pts; ///< user-specified first expected pts, in samples int comp; ///< current resample compensation /* options */ int resample; float min_delta_sec; int max_comp; /* set by filter_frame() 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 #define F AV_OPT_FLAG_FILTERING_PARAM static const AVOption asyncts_options[] = { { "compensate", "Stretch/squeeze the data to make it match the timestamps", OFFSET(resample), AV_OPT_TYPE_BOOL, { .i64 = 0 }, 0, 1, A|F }, { "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, { .dbl = 0.1 }, 0, INT_MAX, A|F }, { "max_comp", "Maximum compensation in samples per second.", OFFSET(max_comp), AV_OPT_TYPE_INT, { .i64 = 500 }, 0, INT_MAX, A|F }, { "first_pts", "Assume the first pts should be this value.", OFFSET(first_pts), AV_OPT_TYPE_INT64, { .i64 = AV_NOPTS_VALUE }, INT64_MIN, INT64_MAX, A|F }, { NULL } }; AVFILTER_DEFINE_CLASS(asyncts); static av_cold int init(AVFilterContext *ctx) { ASyncContext *s = ctx->priv; s->pts = AV_NOPTS_VALUE; s->first_frame = 1; return 0; } static av_cold 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; } /* 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 void handle_trimming(AVFilterContext *ctx) { ASyncContext *s = ctx->priv; if (s->pts < s->first_pts) { int delta = FFMIN(s->first_pts - s->pts, avresample_available(s->avr)); av_log(ctx, AV_LOG_VERBOSE, "Trimming %d samples from start\n", delta); avresample_read(s->avr, NULL, delta); s->pts += delta; } else if (s->first_frame) s->pts = s->first_pts; } static int request_frame(AVFilterLink *link) { AVFilterContext *ctx = link->src; ASyncContext *s = ctx->priv; int ret = 0; int nb_samples; s->got_output = 0; ret = ff_request_frame(ctx->inputs[0]); /* flush the fifo */ if (ret == AVERROR_EOF) { if (s->first_pts != AV_NOPTS_VALUE) handle_trimming(ctx); if (nb_samples = get_delay(s)) { AVFrame *buf = ff_get_audio_buffer(link, nb_samples); if (!buf) return AVERROR(ENOMEM); ret = avresample_convert(s->avr, buf->extended_data, buf->linesize[0], nb_samples, NULL, 0, 0); if (ret <= 0) { av_frame_free(&buf); return (ret < 0) ? ret : AVERROR_EOF; } buf->pts = s->pts; return ff_filter_frame(link, buf); } } return ret; } static int write_to_fifo(ASyncContext *s, AVFrame *buf) { int ret = avresample_convert(s->avr, NULL, 0, 0, buf->extended_data, buf->linesize[0], buf->nb_samples); av_frame_free(&buf); return ret; } static int filter_frame(AVFilterLink *inlink, AVFrame *buf) { AVFilterContext *ctx = inlink->dst; ASyncContext *s = ctx->priv; AVFilterLink *outlink = ctx->outputs[0]; int nb_channels = av_get_channel_layout_nb_channels(buf->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; int64_t new_pts; /* buffer data until we get the next timestamp */ if (s->pts == AV_NOPTS_VALUE || pts == AV_NOPTS_VALUE) { if (pts != AV_NOPTS_VALUE) { s->pts = pts - get_delay(s); } return write_to_fifo(s, buf); } if (s->first_pts != AV_NOPTS_VALUE) { handle_trimming(ctx); if (!avresample_available(s->avr)) 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 (llabs(delta) > s->min_delta || (s->first_frame && delta && s->first_pts != AV_NOPTS_VALUE)) { 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) { // adjust the compensation if delta is non-zero int delay = get_delay(s); int comp = s->comp + av_clip(delta * inlink->sample_rate / delay, -s->max_comp, s->max_comp); if (comp != s->comp) { av_log(ctx, AV_LOG_VERBOSE, "Compensating %d samples per second.\n", comp); if (avresample_set_compensation(s->avr, comp, inlink->sample_rate) == 0) { s->comp = comp; } } } // adjust PTS to avoid monotonicity errors with input PTS jitter pts -= delta; delta = 0; } if (out_size > 0) { AVFrame *buf_out = ff_get_audio_buffer(outlink, out_size); if (!buf_out) { ret = AVERROR(ENOMEM); goto fail; } if (s->first_frame && delta > 0) { int planar = av_sample_fmt_is_planar(buf_out->format); int planes = planar ? nb_channels : 1; int block_size = av_get_bytes_per_sample(buf_out->format) * (planar ? 1 : nb_channels); int ch; av_samples_set_silence(buf_out->extended_data, 0, delta, nb_channels, buf->format); for (ch = 0; ch < planes; ch++) buf_out->extended_data[ch] += delta * block_size; avresample_read(s->avr, buf_out->extended_data, out_size); for (ch = 0; ch < planes; ch++) buf_out->extended_data[ch] -= delta * block_size; } else { avresample_read(s->avr, buf_out->extended_data, out_size); if (delta > 0) { av_samples_set_silence(buf_out->extended_data, out_size - delta, delta, nb_channels, buf->format); } } buf_out->pts = s->pts; ret = ff_filter_frame(outlink, buf_out); if (ret < 0) goto fail; s->got_output = 1; } else if (avresample_available(s->avr)) { 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)); new_pts = pts - avresample_get_delay(s->avr); /* check for s->pts monotonicity */ if (new_pts > s->pts) { s->pts = new_pts; ret = avresample_convert(s->avr, NULL, 0, 0, buf->extended_data, buf->linesize[0], buf->nb_samples); } else { av_log(ctx, AV_LOG_WARNING, "Non-monotonous timestamps, dropping " "whole buffer.\n"); ret = 0; } s->first_frame = 0; fail: av_frame_free(&buf); return ret; } static const AVFilterPad avfilter_af_asyncts_inputs[] = { { .name = "default", .type = AVMEDIA_TYPE_AUDIO, .filter_frame = filter_frame }, { NULL } }; static const AVFilterPad avfilter_af_asyncts_outputs[] = { { .name = "default", .type = AVMEDIA_TYPE_AUDIO, .config_props = config_props, .request_frame = request_frame }, { NULL } }; AVFilter ff_af_asyncts = { .name = "asyncts", .description = NULL_IF_CONFIG_SMALL("Sync audio data to timestamps."), .init = init, .uninit = uninit, .priv_size = sizeof(ASyncContext), .priv_class = &asyncts_class, .inputs = avfilter_af_asyncts_inputs, .outputs = avfilter_af_asyncts_outputs, };