yading@10: /* yading@10: * This file is part of Libav. yading@10: * yading@10: * Libav is free software; you can redistribute it and/or yading@10: * modify it under the terms of the GNU Lesser General Public yading@10: * License as published by the Free Software Foundation; either yading@10: * version 2.1 of the License, or (at your option) any later version. yading@10: * yading@10: * Libav is distributed in the hope that it will be useful, yading@10: * but WITHOUT ANY WARRANTY; without even the implied warranty of yading@10: * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU yading@10: * Lesser General Public License for more details. yading@10: * yading@10: * You should have received a copy of the GNU Lesser General Public yading@10: * License along with Libav; if not, write to the Free Software yading@10: * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA yading@10: */ yading@10: yading@10: #include "libavresample/avresample.h" yading@10: #include "libavutil/audio_fifo.h" yading@10: #include "libavutil/common.h" yading@10: #include "libavutil/mathematics.h" yading@10: #include "libavutil/opt.h" yading@10: #include "libavutil/samplefmt.h" yading@10: yading@10: #include "audio.h" yading@10: #include "avfilter.h" yading@10: #include "internal.h" yading@10: yading@10: typedef struct ASyncContext { yading@10: const AVClass *class; yading@10: yading@10: AVAudioResampleContext *avr; yading@10: int64_t pts; ///< timestamp in samples of the first sample in fifo yading@10: int min_delta; ///< pad/trim min threshold in samples yading@10: int first_frame; ///< 1 until filter_frame() has processed at least 1 frame with a pts != AV_NOPTS_VALUE yading@10: int64_t first_pts; ///< user-specified first expected pts, in samples yading@10: int comp; ///< current resample compensation yading@10: yading@10: /* options */ yading@10: int resample; yading@10: float min_delta_sec; yading@10: int max_comp; yading@10: yading@10: /* set by filter_frame() to signal an output frame to request_frame() */ yading@10: int got_output; yading@10: } ASyncContext; yading@10: yading@10: #define OFFSET(x) offsetof(ASyncContext, x) yading@10: #define A AV_OPT_FLAG_AUDIO_PARAM yading@10: #define F AV_OPT_FLAG_FILTERING_PARAM yading@10: static const AVOption asyncts_options[] = { yading@10: { "compensate", "Stretch/squeeze the data to make it match the timestamps", OFFSET(resample), AV_OPT_TYPE_INT, { .i64 = 0 }, 0, 1, A|F }, yading@10: { "min_delta", "Minimum difference between timestamps and audio data " yading@10: "(in seconds) to trigger padding/trimmin the data.", OFFSET(min_delta_sec), AV_OPT_TYPE_FLOAT, { .dbl = 0.1 }, 0, INT_MAX, A|F }, yading@10: { "max_comp", "Maximum compensation in samples per second.", OFFSET(max_comp), AV_OPT_TYPE_INT, { .i64 = 500 }, 0, INT_MAX, A|F }, yading@10: { "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 }, yading@10: { NULL }, yading@10: }; yading@10: yading@10: AVFILTER_DEFINE_CLASS(asyncts); yading@10: yading@10: static int init(AVFilterContext *ctx) yading@10: { yading@10: ASyncContext *s = ctx->priv; yading@10: yading@10: s->pts = AV_NOPTS_VALUE; yading@10: s->first_frame = 1; yading@10: yading@10: return 0; yading@10: } yading@10: yading@10: static void uninit(AVFilterContext *ctx) yading@10: { yading@10: ASyncContext *s = ctx->priv; yading@10: yading@10: if (s->avr) { yading@10: avresample_close(s->avr); yading@10: avresample_free(&s->avr); yading@10: } yading@10: } yading@10: yading@10: static int config_props(AVFilterLink *link) yading@10: { yading@10: ASyncContext *s = link->src->priv; yading@10: int ret; yading@10: yading@10: s->min_delta = s->min_delta_sec * link->sample_rate; yading@10: link->time_base = (AVRational){1, link->sample_rate}; yading@10: yading@10: s->avr = avresample_alloc_context(); yading@10: if (!s->avr) yading@10: return AVERROR(ENOMEM); yading@10: yading@10: av_opt_set_int(s->avr, "in_channel_layout", link->channel_layout, 0); yading@10: av_opt_set_int(s->avr, "out_channel_layout", link->channel_layout, 0); yading@10: av_opt_set_int(s->avr, "in_sample_fmt", link->format, 0); yading@10: av_opt_set_int(s->avr, "out_sample_fmt", link->format, 0); yading@10: av_opt_set_int(s->avr, "in_sample_rate", link->sample_rate, 0); yading@10: av_opt_set_int(s->avr, "out_sample_rate", link->sample_rate, 0); yading@10: yading@10: if (s->resample) yading@10: av_opt_set_int(s->avr, "force_resampling", 1, 0); yading@10: yading@10: if ((ret = avresample_open(s->avr)) < 0) yading@10: return ret; yading@10: yading@10: return 0; yading@10: } yading@10: yading@10: /* get amount of data currently buffered, in samples */ yading@10: static int64_t get_delay(ASyncContext *s) yading@10: { yading@10: return avresample_available(s->avr) + avresample_get_delay(s->avr); yading@10: } yading@10: yading@10: static void handle_trimming(AVFilterContext *ctx) yading@10: { yading@10: ASyncContext *s = ctx->priv; yading@10: yading@10: if (s->pts < s->first_pts) { yading@10: int delta = FFMIN(s->first_pts - s->pts, avresample_available(s->avr)); yading@10: av_log(ctx, AV_LOG_VERBOSE, "Trimming %d samples from start\n", yading@10: delta); yading@10: avresample_read(s->avr, NULL, delta); yading@10: s->pts += delta; yading@10: } else if (s->first_frame) yading@10: s->pts = s->first_pts; yading@10: } yading@10: yading@10: static int request_frame(AVFilterLink *link) yading@10: { yading@10: AVFilterContext *ctx = link->src; yading@10: ASyncContext *s = ctx->priv; yading@10: int ret = 0; yading@10: int nb_samples; yading@10: yading@10: s->got_output = 0; yading@10: while (ret >= 0 && !s->got_output) yading@10: ret = ff_request_frame(ctx->inputs[0]); yading@10: yading@10: /* flush the fifo */ yading@10: if (ret == AVERROR_EOF) { yading@10: if (s->first_pts != AV_NOPTS_VALUE) yading@10: handle_trimming(ctx); yading@10: yading@10: if (nb_samples = get_delay(s)) { yading@10: AVFrame *buf = ff_get_audio_buffer(link, nb_samples); yading@10: if (!buf) yading@10: return AVERROR(ENOMEM); yading@10: ret = avresample_convert(s->avr, buf->extended_data, yading@10: buf->linesize[0], nb_samples, NULL, 0, 0); yading@10: if (ret <= 0) { yading@10: av_frame_free(&buf); yading@10: return (ret < 0) ? ret : AVERROR_EOF; yading@10: } yading@10: yading@10: buf->pts = s->pts; yading@10: return ff_filter_frame(link, buf); yading@10: } yading@10: } yading@10: yading@10: return ret; yading@10: } yading@10: yading@10: static int write_to_fifo(ASyncContext *s, AVFrame *buf) yading@10: { yading@10: int ret = avresample_convert(s->avr, NULL, 0, 0, buf->extended_data, yading@10: buf->linesize[0], buf->nb_samples); yading@10: av_frame_free(&buf); yading@10: return ret; yading@10: } yading@10: yading@10: static int filter_frame(AVFilterLink *inlink, AVFrame *buf) yading@10: { yading@10: AVFilterContext *ctx = inlink->dst; yading@10: ASyncContext *s = ctx->priv; yading@10: AVFilterLink *outlink = ctx->outputs[0]; yading@10: int nb_channels = av_get_channel_layout_nb_channels(buf->channel_layout); yading@10: int64_t pts = (buf->pts == AV_NOPTS_VALUE) ? buf->pts : yading@10: av_rescale_q(buf->pts, inlink->time_base, outlink->time_base); yading@10: int out_size, ret; yading@10: int64_t delta; yading@10: int64_t new_pts; yading@10: yading@10: /* buffer data until we get the next timestamp */ yading@10: if (s->pts == AV_NOPTS_VALUE || pts == AV_NOPTS_VALUE) { yading@10: if (pts != AV_NOPTS_VALUE) { yading@10: s->pts = pts - get_delay(s); yading@10: } yading@10: return write_to_fifo(s, buf); yading@10: } yading@10: yading@10: if (s->first_pts != AV_NOPTS_VALUE) { yading@10: handle_trimming(ctx); yading@10: if (!avresample_available(s->avr)) yading@10: return write_to_fifo(s, buf); yading@10: } yading@10: yading@10: /* when we have two timestamps, compute how many samples would we have yading@10: * to add/remove to get proper sync between data and timestamps */ yading@10: delta = pts - s->pts - get_delay(s); yading@10: out_size = avresample_available(s->avr); yading@10: yading@10: if (labs(delta) > s->min_delta || yading@10: (s->first_frame && delta && s->first_pts != AV_NOPTS_VALUE)) { yading@10: av_log(ctx, AV_LOG_VERBOSE, "Discontinuity - %"PRId64" samples.\n", delta); yading@10: out_size = av_clipl_int32((int64_t)out_size + delta); yading@10: } else { yading@10: if (s->resample) { yading@10: // adjust the compensation if delta is non-zero yading@10: int delay = get_delay(s); yading@10: int comp = s->comp + av_clip(delta * inlink->sample_rate / delay, yading@10: -s->max_comp, s->max_comp); yading@10: if (comp != s->comp) { yading@10: av_log(ctx, AV_LOG_VERBOSE, "Compensating %d samples per second.\n", comp); yading@10: if (avresample_set_compensation(s->avr, comp, inlink->sample_rate) == 0) { yading@10: s->comp = comp; yading@10: } yading@10: } yading@10: } yading@10: // adjust PTS to avoid monotonicity errors with input PTS jitter yading@10: pts -= delta; yading@10: delta = 0; yading@10: } yading@10: yading@10: if (out_size > 0) { yading@10: AVFrame *buf_out = ff_get_audio_buffer(outlink, out_size); yading@10: if (!buf_out) { yading@10: ret = AVERROR(ENOMEM); yading@10: goto fail; yading@10: } yading@10: yading@10: if (s->first_frame && delta > 0) { yading@10: int ch; yading@10: yading@10: av_samples_set_silence(buf_out->extended_data, 0, delta, yading@10: nb_channels, buf->format); yading@10: yading@10: for (ch = 0; ch < nb_channels; ch++) yading@10: buf_out->extended_data[ch] += delta; yading@10: yading@10: avresample_read(s->avr, buf_out->extended_data, out_size); yading@10: yading@10: for (ch = 0; ch < nb_channels; ch++) yading@10: buf_out->extended_data[ch] -= delta; yading@10: } else { yading@10: avresample_read(s->avr, buf_out->extended_data, out_size); yading@10: yading@10: if (delta > 0) { yading@10: av_samples_set_silence(buf_out->extended_data, out_size - delta, yading@10: delta, nb_channels, buf->format); yading@10: } yading@10: } yading@10: buf_out->pts = s->pts; yading@10: ret = ff_filter_frame(outlink, buf_out); yading@10: if (ret < 0) yading@10: goto fail; yading@10: s->got_output = 1; yading@10: } else if (avresample_available(s->avr)) { yading@10: av_log(ctx, AV_LOG_WARNING, "Non-monotonous timestamps, dropping " yading@10: "whole buffer.\n"); yading@10: } yading@10: yading@10: /* drain any remaining buffered data */ yading@10: avresample_read(s->avr, NULL, avresample_available(s->avr)); yading@10: yading@10: new_pts = pts - avresample_get_delay(s->avr); yading@10: /* check for s->pts monotonicity */ yading@10: if (new_pts > s->pts) { yading@10: s->pts = new_pts; yading@10: ret = avresample_convert(s->avr, NULL, 0, 0, buf->extended_data, yading@10: buf->linesize[0], buf->nb_samples); yading@10: } else { yading@10: av_log(ctx, AV_LOG_WARNING, "Non-monotonous timestamps, dropping " yading@10: "whole buffer.\n"); yading@10: ret = 0; yading@10: } yading@10: yading@10: s->first_frame = 0; yading@10: fail: yading@10: av_frame_free(&buf); yading@10: yading@10: return ret; yading@10: } yading@10: yading@10: static const AVFilterPad avfilter_af_asyncts_inputs[] = { yading@10: { yading@10: .name = "default", yading@10: .type = AVMEDIA_TYPE_AUDIO, yading@10: .filter_frame = filter_frame yading@10: }, yading@10: { NULL } yading@10: }; yading@10: yading@10: static const AVFilterPad avfilter_af_asyncts_outputs[] = { yading@10: { yading@10: .name = "default", yading@10: .type = AVMEDIA_TYPE_AUDIO, yading@10: .config_props = config_props, yading@10: .request_frame = request_frame yading@10: }, yading@10: { NULL } yading@10: }; yading@10: yading@10: AVFilter avfilter_af_asyncts = { yading@10: .name = "asyncts", yading@10: .description = NULL_IF_CONFIG_SMALL("Sync audio data to timestamps"), yading@10: yading@10: .init = init, yading@10: .uninit = uninit, yading@10: yading@10: .priv_size = sizeof(ASyncContext), yading@10: .priv_class = &asyncts_class, yading@10: yading@10: .inputs = avfilter_af_asyncts_inputs, yading@10: .outputs = avfilter_af_asyncts_outputs, yading@10: };