lavfi: add asyncts filter.

doc/filters.texi

@@ -137,6 +137,25 @@ aformat=sample_fmts\=u8\,s16:channel_layouts\=stereo
 
 Pass the audio source unchanged to the output.
 
+@section asyncts
+Synchronize audio data with timestamps by squeezing/stretching it and/or
+dropping samples/adding silence when needed.
+
+The filter accepts the following named parameters:
+@table @option
+
+@item compensate
+Enable stretching/squeezing the data to make it match the timestamps.
+
+@item min_delta
+Minimum difference between timestamps and audio data (in seconds) to trigger
+adding/dropping samples.
+
+@item max_comp
+Maximum compensation in samples per second.
+
+@end table
+
 @section resample
 Convert the audio sample format, sample rate and channel layout. This filter is
 not meant to be used directly, it is inserted automatically by libavfilter

libavfilter/Makefile

@@ -1,5 +1,6 @@
 NAME = avfilter
 FFLIBS = avutil swscale
+FFLIBS-$(CONFIG_ASYNCTS_FILTER) += avresample
 FFLIBS-$(CONFIG_MOVIE_FILTER) += avformat avcodec
 FFLIBS-$(CONFIG_RESAMPLE_FILTER) += avresample
 
@@ -24,6 +25,7 @@ OBJS = allfilters.o                                                     \
 
 OBJS-$(CONFIG_AFORMAT_FILTER)                += af_aformat.o
 OBJS-$(CONFIG_ANULL_FILTER)                  += af_anull.o
+OBJS-$(CONFIG_ASYNCTS_FILTER)                += af_asyncts.o
 OBJS-$(CONFIG_RESAMPLE_FILTER)               += af_resample.o
 
 OBJS-$(CONFIG_ANULLSRC_FILTER)               += asrc_anullsrc.o

libavfilter/af_asyncts.c

@@ -0,0 +1,237 @@
+/*
+ * 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);
+    }
+
+    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 }},
+};

libavfilter/allfilters.c

@@ -36,6 +36,7 @@ void avfilter_register_all(void)
 
     REGISTER_FILTER (AFORMAT,     aformat,     af);
     REGISTER_FILTER (ANULL,       anull,       af);
+    REGISTER_FILTER (ASYNCTS,     asyncts,     af);
     REGISTER_FILTER (RESAMPLE,    resample,    af);
 
     REGISTER_FILTER (ANULLSRC,    anullsrc,    asrc);