Merge commit 'b35e5d985dd12acf9a0aaa52334134edcf35d68e'

* commit 'b35e5d985dd12acf9a0aaa52334134edcf35d68e':
  doc: improve documentation for the asyncts filter first_pts option
  asyncts: fix the asyncts behavior when using the first_pts option

Conflicts:
	libavfilter/af_asyncts.c

Merged-by: Michael Niedermayer <michaelni@gmx.at>

doc/filters.texi

@@ -782,11 +782,12 @@ Maximum compensation in samples per second. Relevant only with compensate=1.
 Default value 500.
 
 @item first_pts
-Assume the first pts should be this value.
+Assume the first pts should be this value. The time base is 1 / sample rate.
 This allows for padding/trimming at the start of stream. By default, no
 assumption is made about the first frame's expected pts, so no padding or
 trimming is done. For example, this could be set to 0 to pad the beginning with
-silence if an audio stream starts after the video stream.
+silence if an audio stream starts after the video stream or to trim any samples
+with a negative pts due to encoder delay.
 
 @end table
 

libavfilter/af_asyncts.c

@@ -33,6 +33,8 @@ typedef struct ASyncContext {
     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
 
     /* options */
     int resample;
@@ -51,7 +53,7 @@ static const AVOption asyncts_options[] = {
     { "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(pts),           AV_OPT_TYPE_INT64, { .i64 = AV_NOPTS_VALUE }, INT64_MIN, INT64_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 },
 };
 
@@ -69,6 +71,9 @@ static int init(AVFilterContext *ctx, const char *args)
         return ret;
     av_opt_free(s);
 
+    s->pts         = AV_NOPTS_VALUE;
+    s->first_frame = 1;
+
     return 0;
 }
 
@@ -116,6 +121,20 @@ 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;
@@ -128,7 +147,11 @@ static int request_frame(AVFilterLink *link)
         ret = ff_request_frame(ctx->inputs[0]);
 
     /* flush the fifo */
-    if (ret == AVERROR_EOF && (nb_samples = get_delay(s))) {
+    if (ret == AVERROR_EOF) {
+        if (s->first_pts != AV_NOPTS_VALUE)
+            handle_trimming(ctx);
+
+        if (nb_samples = get_delay(s)) {
         AVFilterBufferRef *buf = ff_get_audio_buffer(link, AV_PERM_WRITE,
                                                      nb_samples);
         if (!buf)
@@ -143,6 +166,7 @@ static int request_frame(AVFilterLink *link)
         buf->pts = s->pts;
         return ff_filter_frame(link, buf);
         }
+    }
 
     return ret;
 }
@@ -179,12 +203,18 @@ static int filter_frame(AVFilterLink *inlink, AVFilterBufferRef *buf)
         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 (labs(delta) > s->min_delta) {
+    if (labs(delta) > s->min_delta || (s->first_frame && delta)) {
         av_log(ctx, AV_LOG_VERBOSE, "Discontinuity - %"PRId64" samples.\n", delta);
         out_size = av_clipl_int32((int64_t)out_size + delta);
     } else {
@@ -204,18 +234,33 @@ static int filter_frame(AVFilterLink *inlink, AVFilterBufferRef *buf)
             goto fail;
         }
 
+        if (s->first_frame && delta > 0) {
+            int ch;
+
+            av_samples_set_silence(buf_out->extended_data, 0, delta,
+                                   nb_channels, buf->format);
+
+            for (ch = 0; ch < nb_channels; ch++)
+                buf_out->extended_data[ch] += delta;
+
+            avresample_read(s->avr, buf_out->extended_data, out_size);
+
+            for (ch = 0; ch < nb_channels; ch++)
+                buf_out->extended_data[ch] -= delta;
+        } else {
             avresample_read(s->avr, 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);
             }
+        }
+        buf_out->pts = s->pts;
         ret = ff_filter_frame(outlink, buf_out);
         if (ret < 0)
             goto fail;
         s->got_output = 1;
-    } else {
+    } else if (avresample_available(s->avr)) {
         av_log(ctx, AV_LOG_WARNING, "Non-monotonous timestamps, dropping "
                "whole buffer.\n");
     }
@@ -227,6 +272,7 @@ static int filter_frame(AVFilterLink *inlink, AVFilterBufferRef *buf)
     ret = avresample_convert(s->avr, NULL, 0, 0, buf->extended_data,
                              buf->linesize[0], buf->audio->nb_samples);
 
+    s->first_frame = 0;
 fail:
     avfilter_unref_buffer(buf);