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