asyncts first_pts

gistfile1.diff

From a0bfc6fd8e28d0d378c537f38ea48bbbd5aea96a Mon Sep 17 00:00:00 2001
From: Justin Ruggles <[email protected]>
Date: Tue, 11 Dec 2012 17:36:09 -0500
Subject: [PATCH] WIP: fix the asyncts first_pts option

---
 libavfilter/af_asyncts.c |   79 ++++++++++++++++++++++++++++++++++-----------
 libavfilter/buffersrc.c  |    7 ++++
 2 files changed, 66 insertions(+), 20 deletions(-)

diff --git a/libavfilter/af_asyncts.c b/libavfilter/af_asyncts.c
index 5d009f0..3630a12 100644
--- a/libavfilter/af_asyncts.c
+++ b/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;        ///< 0 if filter_frame() got at least 1 frame with a pts != AV_NOPTS_VALUE
+    int drop_samples;
 
     /* options */
     int resample;
@@ -75,6 +77,8 @@ static int init(AVFilterContext *ctx, const char *args)
     }
     av_opt_free(s);
 
+    s->first_frame = 1;
+
     return 0;
 }
 
@@ -116,6 +120,12 @@ static int config_props(AVFilterLink *link)
     return 0;
 }
 
+/* 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 int request_frame(AVFilterLink *link)
 {
     AVFilterContext *ctx = link->src;
@@ -128,7 +138,7 @@ static int request_frame(AVFilterLink *link)
         ret = ff_request_frame(ctx->inputs[0]);
 
     /* flush the fifo */
-    if (ret == AVERROR_EOF && (nb_samples = avresample_get_delay(s->avr))) {
+    if (ret == AVERROR_EOF && (nb_samples = get_delay(s))) {
         AVFilterBufferRef *buf = ff_get_audio_buffer(link, AV_PERM_WRITE,
                                                      nb_samples);
         if (!buf)
@@ -140,6 +150,7 @@ static int request_frame(AVFilterLink *link)
             return (ret < 0) ? ret : AVERROR_EOF;
         }
 
+        av_log(0,0,"filter_frame: %d samples\n", nb_samples);
         buf->pts = s->pts;
         return ff_filter_frame(link, buf);
     }
@@ -155,12 +166,6 @@ static int write_to_fifo(ASyncContext *s, AVFilterBufferRef *buf)
     return ret;
 }
 
-/* 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 int filter_frame(AVFilterLink *inlink, AVFilterBufferRef *buf)
 {
     AVFilterContext  *ctx = inlink->dst;
@@ -173,12 +178,8 @@ static int filter_frame(AVFilterLink *inlink, AVFilterBufferRef *buf)
     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);
-        }
-        return write_to_fifo(s, buf);
-    }
+    if (s->pts == AV_NOPTS_VALUE && pts != AV_NOPTS_VALUE)
+        s->pts = pts - get_delay(s);
 
     /* now wait for the next timestamp */
     if (pts == AV_NOPTS_VALUE) {
@@ -187,10 +188,20 @@ static int filter_frame(AVFilterLink *inlink, AVFilterBufferRef *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);
+    delta = pts - s->pts - get_delay(s);
+    if (s->first_frame && delta < 0) {
+        int buffered_samples = avresample_available(s->avr);
+        s->drop_samples = -delta;
+        if (buffered_samples) {
+            int drain = FFMIN(s->drop_samples, buffered_samples);
+            avresample_read(s->avr, NULL, drain);
+            s->drop_samples -= drain;
+            delta += drain;
+        }
+    }
     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 {
@@ -202,6 +213,10 @@ static int filter_frame(AVFilterLink *inlink, AVFilterBufferRef *buf)
         delta = 0;
     }
 
+    if (s->first_frame)
+        av_log(0,0,"first_pts=%ld pts=%ld delta=%ld avail=%d out_size=%d\n",
+               s->pts, pts, delta, avresample_available(s->avr), out_size);
+
     if (out_size > 0) {
         AVFilterBufferRef *buf_out = ff_get_audio_buffer(outlink, AV_PERM_WRITE,
                                                          out_size);
@@ -210,13 +225,30 @@ static int filter_frame(AVFilterLink *inlink, AVFilterBufferRef *buf)
             goto fail;
         }
 
-        avresample_read(s->avr, buf_out->extended_data, out_size);
-        buf_out->pts = s->pts;
+        if (s->first_frame && delta > 0) {
+            int ch;
+            uint8_t **data0 = av_mallocz(nb_channels * sizeof(*data0));
+            if (!data0)
+                return AVERROR(ENOMEM);
 
-        if (delta > 0) {
-            av_samples_set_silence(buf_out->extended_data, out_size - delta,
-                                   delta, nb_channels, buf->format);
+            av_samples_set_silence(buf_out->extended_data, 0, delta,
+                                   nb_channels, buf->format);
+
+            for (ch = 0; ch < nb_channels; ch++)
+                data0[ch] = buf_out->extended_data[ch] + delta;
+            avresample_read(s->avr, data0, out_size);
+
+            free(data0);
+        } else {
+            avresample_read(s->avr, buf_out->extended_data, out_size);
+
+            if (delta > 0) {
+                av_samples_set_silence(buf_out->extended_data, out_size - delta,
+                                       delta, nb_channels, buf->format);
+            }
         }
+        av_log(0,0,"filter_frame: %d samples\n", out_size);
+        buf_out->pts = s->pts;
         ret = ff_filter_frame(outlink, buf_out);
         if (ret < 0)
             goto fail;
@@ -233,6 +265,13 @@ 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);
 
+    if (s->drop_samples > 0) {
+        int drain = FFMIN(s->drop_samples, avresample_available(s->avr));
+        avresample_read(s->avr, NULL, drain);
+        s->drop_samples -= drain;
+    }
+
+    s->first_frame = 0;
 fail:
     avfilter_unref_buffer(buf);
 
diff --git a/libavfilter/buffersrc.c b/libavfilter/buffersrc.c
index 3cee68d..dc5e08e 100644
--- a/libavfilter/buffersrc.c
+++ b/libavfilter/buffersrc.c
@@ -54,6 +54,7 @@ typedef struct {
     char    *channel_layout_str;
 
     int eof;
+    int first;
 } BufferSourceContext;
 
 #define CHECK_VIDEO_PARAM_CHANGE(s, c, width, height, format)\
@@ -246,6 +247,7 @@ static av_cold int init_audio(AVFilterContext *ctx, const char *args)
            "ch layout:%s\n", s->time_base.num, s->time_base.den, s->sample_fmt_str,
            s->sample_rate, s->channel_layout_str);
 
+    s->first = 2;
 fail:
     av_opt_free(s);
     return ret;
@@ -327,6 +329,11 @@ static int request_frame(AVFilterLink *link)
     }
     av_fifo_generic_read(c->fifo, &buf, sizeof(buf), NULL);
 
+    if (c->first) {
+        buf->pts = AV_NOPTS_VALUE;
+        c->first--;
+    }
+
     ff_filter_frame(link, buf);
 
     return ret;
-- 
1.7.1