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							- /*
 -  * 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/mathematics.h"
 - #include "libavutil/opt.h"
 - #include "libavutil/samplefmt.h"
 - 
 - #include "audio.h"
 - #include "avfilter.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;
 - } 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 },
 -     { 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, void *opaque)
 - {
 -     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);
 - 
 -     s->pts = AV_NOPTS_VALUE;
 - 
 -     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 = avfilter_request_frame(ctx->inputs[0]);
 -     int nb_samples;
 - 
 -     /* 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;
 -         ff_filter_samples(link, buf);
 -         return 0;
 -     }
 - 
 -     return ret;
 - }
 - 
 - static void write_to_fifo(ASyncContext *s, AVFilterBufferRef *buf)
 - {
 -     avresample_convert(s->avr, NULL, 0, 0, (void**)buf->extended_data,
 -                        buf->linesize[0], buf->audio->nb_samples);
 -     avfilter_unref_buffer(buf);
 - }
 - 
 - /* 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 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;
 -     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);
 -         }
 -         write_to_fifo(s, buf);
 -         return;
 -     }
 - 
 -     /* now wait for the next timestamp */
 -     if (pts == AV_NOPTS_VALUE) {
 -         write_to_fifo(s, buf);
 -         return;
 -     }
 - 
 -     /* 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 += 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)
 -             return;
 - 
 -         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);
 -         }
 -         ff_filter_samples(outlink, buf_out);
 -     } 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);
 -     avresample_convert(s->avr, NULL, 0, 0, (void**)buf->extended_data,
 -                        buf->linesize[0], buf->audio->nb_samples);
 -     avfilter_unref_buffer(buf);
 - }
 - 
 - 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 }},
 - };
 
 
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