audio: Use GST_BUFFER_PTS instead of deprecated GST_BUFFER_TIMESTAMP

GST_BUFFER_PTS already used in audio code base (e.g. gstaudiodecoder),
so migrate completely from deprecated GST_BUFFER_TIMESTAMP for better
readability, as gstcompat.h defines GST_BUFFER_TIMESTAMP directly to PTS
anyway.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1048>

diff --git a/gst-libs/gst/audio/audio.c b/gst-libs/gst/audio/audio.c
index 075b8db..c4aac4d 100644 (file)
--- a/gst-libs/gst/audio/audio.c
+++ b/gst-libs/gst/audio/audio.c
@@ -91,7 +91,7 @@ gst_audio_buffer_clip (GstBuffer * buffer, const GstSegment * segment,
       segment->format == GST_FORMAT_DEFAULT, buffer);
   g_return_val_if_fail (GST_IS_BUFFER (buffer), NULL);
 
-  if (!GST_BUFFER_TIMESTAMP_IS_VALID (buffer))
+  if (!GST_BUFFER_PTS_IS_VALID (buffer))
     /* No timestamp - assume the buffer is completely in the segment */
     return buffer;
 
@@ -109,7 +109,7 @@ gst_audio_buffer_clip (GstBuffer * buffer, const GstSegment * segment,
   if (!size)
     return buffer;
 
-  timestamp = GST_BUFFER_TIMESTAMP (buffer);
+  timestamp = GST_BUFFER_PTS (buffer);
   GST_DEBUG ("timestamp %" GST_TIME_FORMAT, GST_TIME_ARGS (timestamp));
   if (GST_BUFFER_DURATION_IS_VALID (buffer)) {
     duration = GST_BUFFER_DURATION (buffer);
@@ -214,9 +214,9 @@ gst_audio_buffer_clip (GstBuffer * buffer, const GstSegment * segment,
   if (trim == 0 && size == osize) {
     ret = buffer;
 
-    if (GST_BUFFER_TIMESTAMP (ret) != timestamp) {
+    if (GST_BUFFER_PTS (ret) != timestamp) {
       ret = gst_buffer_make_writable (ret);
-      GST_BUFFER_TIMESTAMP (ret) = timestamp;
+      GST_BUFFER_PTS (ret) = timestamp;
     }
     if (GST_BUFFER_DURATION (ret) != duration) {
       ret = gst_buffer_make_writable (ret);
@@ -229,7 +229,7 @@ gst_audio_buffer_clip (GstBuffer * buffer, const GstSegment * segment,
 
     GST_DEBUG ("timestamp %" GST_TIME_FORMAT, GST_TIME_ARGS (timestamp));
     if (ret) {
-      GST_BUFFER_TIMESTAMP (ret) = timestamp;
+      GST_BUFFER_PTS (ret) = timestamp;
 
       if (change_duration)
         GST_BUFFER_DURATION (ret) = duration;

diff --git a/gst-libs/gst/audio/gstaudiobasesink.c b/gst-libs/gst/audio/gstaudiobasesink.c
index 7ce15cd..359b1b0 100644 (file)
--- a/gst-libs/gst/audio/gstaudiobasesink.c
+++ b/gst-libs/gst/audio/gstaudiobasesink.c
@@ -1864,7 +1864,7 @@ gst_audio_base_sink_render (GstBaseSink * bsink, GstBuffer * buf)
 
   samples = size / bpf;
 
-  time = GST_BUFFER_TIMESTAMP (buf);
+  time = GST_BUFFER_PTS (buf);
 
   /* Last ditch attempt to ensure that we only play silence if
    * we are in trickmode no-audio mode (or if a buffer is marked as a GAP)

diff --git a/gst-libs/gst/audio/gstaudiobasesrc.c b/gst-libs/gst/audio/gstaudiobasesrc.c
index b30a357..3df095e 100644 (file)
--- a/gst-libs/gst/audio/gstaudiobasesrc.c
+++ b/gst-libs/gst/audio/gstaudiobasesrc.c
@@ -1027,7 +1027,7 @@ gst_audio_base_src_create (GstBaseSrc * bsrc, guint64 offset, guint length,
 no_sync:
   GST_OBJECT_UNLOCK (src);
 
-  GST_BUFFER_TIMESTAMP (buf) = timestamp;
+  GST_BUFFER_PTS (buf) = timestamp;
   GST_BUFFER_DURATION (buf) = duration;
   GST_BUFFER_OFFSET (buf) = sample;
   GST_BUFFER_OFFSET_END (buf) = sample + samples;
@@ -1035,7 +1035,7 @@ no_sync:
   *outbuf = buf;
 
   GST_LOG_OBJECT (src, "Pushed buffer timestamp %" GST_TIME_FORMAT,
-      GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)));
+      GST_TIME_ARGS (GST_BUFFER_PTS (buf)));
 
   return GST_FLOW_OK;
 

diff --git a/gst-libs/gst/audio/gstaudiocdsrc.c b/gst-libs/gst/audio/gstaudiocdsrc.c
index 81421d3..d295f39 100644 (file)
--- a/gst-libs/gst/audio/gstaudiocdsrc.c
+++ b/gst-libs/gst/audio/gstaudiocdsrc.c
@@ -1764,7 +1764,7 @@ gst_audio_cd_src_create (GstPushSrc * pushsrc, GstBuffer ** buffer)
         GST_SECOND, 44100);
   }
 
-  GST_BUFFER_TIMESTAMP (buf) = position;
+  GST_BUFFER_PTS (buf) = position;
   GST_BUFFER_DURATION (buf) = duration;
 
   GST_LOG_OBJECT (src, "pushing sector %d with timestamp %" GST_TIME_FORMAT,

diff --git a/gst-libs/gst/audio/gstaudiodecoder.c b/gst-libs/gst/audio/gstaudiodecoder.c
index 6742948..00f28a9 100644 (file)
--- a/gst-libs/gst/audio/gstaudiodecoder.c
+++ b/gst-libs/gst/audio/gstaudiodecoder.c
@@ -978,12 +978,12 @@ gst_audio_decoder_push_forward (GstAudioDecoder * dec, GstBuffer * buf)
   }
 
   ctx->had_output_data = TRUE;
-  ts = GST_BUFFER_TIMESTAMP (buf);
+  ts = GST_BUFFER_PTS (buf);
 
   GST_LOG_OBJECT (dec,
       "clipping buffer of size %" G_GSIZE_FORMAT " with ts %" GST_TIME_FORMAT
       ", duration %" GST_TIME_FORMAT, gst_buffer_get_size (buf),
-      GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)),
+      GST_TIME_ARGS (GST_BUFFER_PTS (buf)),
       GST_TIME_ARGS (GST_BUFFER_DURATION (buf)));
 
   /* clip buffer */
@@ -1012,11 +1012,11 @@ gst_audio_decoder_push_forward (GstAudioDecoder * dec, GstBuffer * buf)
   }
 
   /* track where we are */
-  if (G_LIKELY (GST_BUFFER_TIMESTAMP_IS_VALID (buf))) {
+  if (G_LIKELY (GST_BUFFER_PTS_IS_VALID (buf))) {
     /* duration should always be valid for raw audio */
     g_assert (GST_BUFFER_DURATION_IS_VALID (buf));
     dec->output_segment.position =
-        GST_BUFFER_TIMESTAMP (buf) + GST_BUFFER_DURATION (buf);
+        GST_BUFFER_PTS (buf) + GST_BUFFER_DURATION (buf);
   }
 
   if (klass->pre_push) {
@@ -1034,7 +1034,7 @@ gst_audio_decoder_push_forward (GstAudioDecoder * dec, GstBuffer * buf)
   GST_LOG_OBJECT (dec,
       "pushing buffer of size %" G_GSIZE_FORMAT " with ts %" GST_TIME_FORMAT
       ", duration %" GST_TIME_FORMAT, gst_buffer_get_size (buf),
-      GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)),
+      GST_TIME_ARGS (GST_BUFFER_PTS (buf)),
       GST_TIME_ARGS (GST_BUFFER_DURATION (buf)));
 
   ret = gst_pad_push (dec->srcpad, buf);
@@ -1061,7 +1061,7 @@ gst_audio_decoder_output (GstAudioDecoder * dec, GstBuffer * buf)
     GST_LOG_OBJECT (dec,
         "output buffer of size %" G_GSIZE_FORMAT " with ts %" GST_TIME_FORMAT
         ", duration %" GST_TIME_FORMAT, gst_buffer_get_size (buf),
-        GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)),
+        GST_TIME_ARGS (GST_BUFFER_PTS (buf)),
         GST_TIME_ARGS (GST_BUFFER_DURATION (buf)));
   }
 
@@ -1079,9 +1079,9 @@ again:
       /* forcibly send current */
       assemble = TRUE;
       GST_LOG_OBJECT (dec, "forcing fragment flush");
-    } else if (av && (!GST_BUFFER_TIMESTAMP_IS_VALID (buf) ||
+    } else if (av && (!GST_BUFFER_PTS_IS_VALID (buf) ||
             !GST_CLOCK_TIME_IS_VALID (priv->out_ts) ||
-            ((diff = GST_CLOCK_DIFF (GST_BUFFER_TIMESTAMP (buf),
+            ((diff = GST_CLOCK_DIFF (GST_BUFFER_PTS (buf),
                         priv->out_ts + priv->out_dur)) > tol) || diff < -tol)) {
       assemble = TRUE;
       GST_LOG_OBJECT (dec, "buffer %d ms apart from current fragment",
@@ -1090,7 +1090,7 @@ again:
       /* add or start collecting */
       if (!av) {
         GST_LOG_OBJECT (dec, "starting new fragment");
-        priv->out_ts = GST_BUFFER_TIMESTAMP (buf);
+        priv->out_ts = GST_BUFFER_PTS (buf);
       } else {
         GST_LOG_OBJECT (dec, "adding to fragment");
       }
@@ -1105,7 +1105,7 @@ again:
       GST_LOG_OBJECT (dec, "assembling fragment");
       inbuf = buf;
       buf = gst_adapter_take_buffer (priv->adapter_out, av);
-      GST_BUFFER_TIMESTAMP (buf) = priv->out_ts;
+      GST_BUFFER_PTS (buf) = priv->out_ts;
       GST_BUFFER_DURATION (buf) = priv->out_dur;
       priv->out_ts = GST_CLOCK_TIME_NONE;
       priv->out_dur = 0;
@@ -1420,7 +1420,7 @@ gst_audio_decoder_finish_frame_or_subframe (GstAudioDecoder * dec,
   }
 
   if (G_LIKELY (priv->frames.length))
-    ts = GST_BUFFER_TIMESTAMP (priv->frames.head->data);
+    ts = GST_BUFFER_PTS (priv->frames.head->data);
   else
     ts = GST_CLOCK_TIME_NONE;
 
@@ -1499,14 +1499,14 @@ gst_audio_decoder_finish_frame_or_subframe (GstAudioDecoder * dec,
 
   buf = gst_buffer_make_writable (buf);
   if (G_LIKELY (GST_CLOCK_TIME_IS_VALID (priv->base_ts))) {
-    GST_BUFFER_TIMESTAMP (buf) =
+    GST_BUFFER_PTS (buf) =
         priv->base_ts +
         GST_FRAMES_TO_CLOCK_TIME (priv->samples, ctx->info.rate);
     GST_BUFFER_DURATION (buf) = priv->base_ts +
         GST_FRAMES_TO_CLOCK_TIME (priv->samples + samples, ctx->info.rate) -
-        GST_BUFFER_TIMESTAMP (buf);
+        GST_BUFFER_PTS (buf);
   } else {
-    GST_BUFFER_TIMESTAMP (buf) = GST_CLOCK_TIME_NONE;
+    GST_BUFFER_PTS (buf) = GST_CLOCK_TIME_NONE;
     GST_BUFFER_DURATION (buf) =
         GST_FRAMES_TO_CLOCK_TIME (samples, ctx->info.rate);
   }
@@ -1624,7 +1624,7 @@ gst_audio_decoder_handle_frame (GstAudioDecoder * dec,
     /* keep around for admin */
     GST_LOG_OBJECT (dec,
         "tracking frame size %" G_GSIZE_FORMAT ", ts %" GST_TIME_FORMAT, size,
-        GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buffer)));
+        GST_TIME_ARGS (GST_BUFFER_PTS (buffer)));
     g_queue_push_tail (&dec->priv->frames, buffer);
     dec->priv->ctx.delay = dec->priv->frames.length;
     GST_OBJECT_LOCK (dec);
@@ -1718,7 +1718,7 @@ gst_audio_decoder_push_buffers (GstAudioDecoder * dec, gboolean force)
       }
       buffer = gst_adapter_take_buffer (priv->adapter, len);
       buffer = gst_buffer_make_writable (buffer);
-      GST_BUFFER_TIMESTAMP (buffer) = ts;
+      GST_BUFFER_PTS (buffer) = ts;
       flush += len;
       priv->force = FALSE;
     } else {
@@ -1952,7 +1952,7 @@ gst_audio_decoder_flush_decode (GstAudioDecoder * dec)
     GstBuffer *buf = GST_BUFFER_CAST (walk->data);
 
     GST_DEBUG_OBJECT (dec, "decoding buffer %p, ts %" GST_TIME_FORMAT,
-        buf, GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)));
+        buf, GST_TIME_ARGS (GST_BUFFER_PTS (buf)));
 
     next = g_list_next (walk);
     /* decode buffer, resulting data prepended to output queue */
@@ -1993,13 +1993,13 @@ gst_audio_decoder_flush_decode (GstAudioDecoder * dec)
         timestamp = 0;
     }
 
-    if (!GST_BUFFER_TIMESTAMP_IS_VALID (buf)) {
+    if (!GST_BUFFER_PTS_IS_VALID (buf)) {
       GST_LOG_OBJECT (dec, "applying reverse interpolated ts %"
           GST_TIME_FORMAT, GST_TIME_ARGS (timestamp));
-      GST_BUFFER_TIMESTAMP (buf) = timestamp;
+      GST_BUFFER_PTS (buf) = timestamp;
     } else {
       /* track otherwise */
-      timestamp = GST_BUFFER_TIMESTAMP (buf);
+      timestamp = GST_BUFFER_PTS (buf);
       GST_LOG_OBJECT (dec, "tracking ts %" GST_TIME_FORMAT,
           GST_TIME_ARGS (timestamp));
     }
@@ -2007,7 +2007,7 @@ gst_audio_decoder_flush_decode (GstAudioDecoder * dec)
     if (G_LIKELY (res == GST_FLOW_OK)) {
       GST_DEBUG_OBJECT (dec, "pushing buffer %p of size %" G_GSIZE_FORMAT ", "
           "time %" GST_TIME_FORMAT ", dur %" GST_TIME_FORMAT, buf,
-          gst_buffer_get_size (buf), GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)),
+          gst_buffer_get_size (buf), GST_TIME_ARGS (GST_BUFFER_PTS (buf)),
           GST_TIME_ARGS (GST_BUFFER_DURATION (buf)));
       /* should be already, but let's be sure */
       buf = gst_buffer_make_writable (buf);
@@ -2050,7 +2050,7 @@ gst_audio_decoder_chain_reverse (GstAudioDecoder * dec, GstBuffer * buf)
   if (G_LIKELY (buf)) {
     GST_DEBUG_OBJECT (dec, "gathering buffer %p of size %" G_GSIZE_FORMAT ", "
         "time %" GST_TIME_FORMAT ", dur %" GST_TIME_FORMAT, buf,
-        gst_buffer_get_size (buf), GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)),
+        gst_buffer_get_size (buf), GST_TIME_ARGS (GST_BUFFER_PTS (buf)),
         GST_TIME_ARGS (GST_BUFFER_DURATION (buf)));
 
     /* add buffer to gather queue */
@@ -2071,7 +2071,7 @@ gst_audio_decoder_chain (GstPad * pad, GstObject * parent, GstBuffer * buffer)
   GST_LOG_OBJECT (dec,
       "received buffer of size %" G_GSIZE_FORMAT " with ts %" GST_TIME_FORMAT
       ", duration %" GST_TIME_FORMAT, gst_buffer_get_size (buffer),
-      GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buffer)),
+      GST_TIME_ARGS (GST_BUFFER_PTS (buffer)),
       GST_TIME_ARGS (GST_BUFFER_DURATION (buffer)));
 
   GST_AUDIO_DECODER_STREAM_LOCK (dec);
@@ -2096,8 +2096,7 @@ gst_audio_decoder_chain (GstPad * pad, GstObject * parent, GstBuffer * buffer)
     /* buffer may claim DISCONT loudly, if it can't tell us where we are now,
      * we'll stick to where we were ...
      * Particularly useful/needed for upstream BYTE based */
-    if (dec->input_segment.rate > 0.0
-        && !GST_BUFFER_TIMESTAMP_IS_VALID (buffer)) {
+    if (dec->input_segment.rate > 0.0 && !GST_BUFFER_PTS_IS_VALID (buffer)) {
       GST_DEBUG_OBJECT (dec, "... but restoring previous ts tracking");
       dec->priv->base_ts = ts;
       dec->priv->samples = samples;
@@ -2293,7 +2292,7 @@ gst_audio_decoder_handle_gap (GstAudioDecoder * dec, GstEvent * event)
 
     /* hand subclass empty frame with duration that needs covering */
     buf = gst_buffer_new ();
-    GST_BUFFER_TIMESTAMP (buf) = timestamp;
+    GST_BUFFER_PTS (buf) = timestamp;
     GST_BUFFER_DURATION (buf) = duration;
     /* best effort, not much error handling */
     gst_audio_decoder_handle_frame (dec, klass, buf);

diff --git a/gst-libs/gst/audio/gstaudioencoder.c b/gst-libs/gst/audio/gstaudioencoder.c
index 2a5192c..4a14d0c 100644 (file)
--- a/gst-libs/gst/audio/gstaudioencoder.c
+++ b/gst-libs/gst/audio/gstaudioencoder.c
@@ -937,16 +937,16 @@ gst_audio_encoder_finish_frame (GstAudioEncoder * enc, GstBuffer * buf,
       /* FIXME ? lookahead could lead to weird ts and duration ?
        * (particularly if not in perfect mode) */
       /* mind sample rounding and produce perfect output */
-      GST_BUFFER_TIMESTAMP (buf) = priv->base_ts +
+      GST_BUFFER_PTS (buf) = priv->base_ts +
           gst_util_uint64_scale (priv->samples - ctx->lookahead, GST_SECOND,
           ctx->info.rate);
-      GST_BUFFER_DTS (buf) = GST_BUFFER_TIMESTAMP (buf);
+      GST_BUFFER_DTS (buf) = GST_BUFFER_PTS (buf);
       GST_DEBUG_OBJECT (enc, "out samples %d", samples);
       if (G_LIKELY (samples > 0)) {
         priv->samples += samples;
         GST_BUFFER_DURATION (buf) = priv->base_ts +
             gst_util_uint64_scale (priv->samples - ctx->lookahead, GST_SECOND,
-            ctx->info.rate) - GST_BUFFER_TIMESTAMP (buf);
+            ctx->info.rate) - GST_BUFFER_PTS (buf);
         priv->last_duration = GST_BUFFER_DURATION (buf);
       } else {
         /* duration forecast in case of handling remainder;
@@ -1008,7 +1008,7 @@ gst_audio_encoder_finish_frame (GstAudioEncoder * enc, GstBuffer * buf,
     GST_LOG_OBJECT (enc,
         "pushing buffer of size %" G_GSIZE_FORMAT " with ts %" GST_TIME_FORMAT
         ", duration %" GST_TIME_FORMAT, size,
-        GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)),
+        GST_TIME_ARGS (GST_BUFFER_PTS (buf)),
         GST_TIME_ARGS (GST_BUFFER_DURATION (buf)));
 
     ret = gst_pad_push (enc->srcpad, buf);
@@ -1236,7 +1236,7 @@ gst_audio_encoder_chain (GstPad * pad, GstObject * parent, GstBuffer * buffer)
   GST_LOG_OBJECT (enc,
       "received buffer of size %" G_GSIZE_FORMAT " with ts %" GST_TIME_FORMAT
       ", duration %" GST_TIME_FORMAT, size,
-      GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buffer)),
+      GST_TIME_ARGS (GST_BUFFER_PTS (buffer)),
       GST_TIME_ARGS (GST_BUFFER_DURATION (buffer)));
 
   /* input should be whole number of sample frames */
@@ -1282,11 +1282,11 @@ gst_audio_encoder_chain (GstPad * pad, GstObject * parent, GstBuffer * buffer)
   GST_LOG_OBJECT (enc,
       "buffer after segment clipping has size %" G_GSIZE_FORMAT " with ts %"
       GST_TIME_FORMAT ", duration %" GST_TIME_FORMAT, size,
-      GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buffer)),
+      GST_TIME_ARGS (GST_BUFFER_PTS (buffer)),
       GST_TIME_ARGS (GST_BUFFER_DURATION (buffer)));
 
   if (!GST_CLOCK_TIME_IS_VALID (priv->base_ts)) {
-    priv->base_ts = GST_BUFFER_TIMESTAMP (buffer);
+    priv->base_ts = GST_BUFFER_PTS (buffer);
     GST_DEBUG_OBJECT (enc, "new base ts %" GST_TIME_FORMAT,
         GST_TIME_ARGS (priv->base_ts));
     gst_audio_encoder_set_base_gp (enc);
@@ -1298,7 +1298,7 @@ gst_audio_encoder_chain (GstPad * pad, GstObject * parent, GstBuffer * buffer)
     GstClockTimeDiff diff = 0;
     GstClockTime next_ts = 0;
 
-    if (GST_BUFFER_TIMESTAMP_IS_VALID (buffer) &&
+    if (GST_BUFFER_PTS_IS_VALID (buffer) &&
         GST_CLOCK_TIME_IS_VALID (priv->base_ts)) {
       guint64 samples;
 
@@ -1310,7 +1310,7 @@ gst_audio_encoder_chain (GstPad * pad, GstObject * parent, GstBuffer * buffer)
           " samples past base_ts %" GST_TIME_FORMAT
           ", expected ts %" GST_TIME_FORMAT, samples,
           GST_TIME_ARGS (priv->base_ts), GST_TIME_ARGS (next_ts));
-      diff = GST_CLOCK_DIFF (next_ts, GST_BUFFER_TIMESTAMP (buffer));
+      diff = GST_CLOCK_DIFF (next_ts, GST_BUFFER_PTS (buffer));
       GST_LOG_OBJECT (enc, "ts diff %d ms", (gint) (diff / GST_MSECOND));
       /* if within tolerance,
        * discard buffer ts and carry on producing perfect stream,
@@ -1339,7 +1339,7 @@ gst_audio_encoder_chain (GstPad * pad, GstObject * parent, GstBuffer * buffer)
         buffer = gst_buffer_make_writable (buffer);
         gst_buffer_resize (buffer, diff_bytes, size - diff_bytes);
 
-        GST_BUFFER_TIMESTAMP (buffer) += diff;
+        GST_BUFFER_PTS (buffer) += diff;
         /* care even less about duration after this */
       } else {
         /* drain stuff prior to resync */
@@ -1352,13 +1352,13 @@ gst_audio_encoder_chain (GstPad * pad, GstObject * parent, GstBuffer * buffer)
           gst_util_uint64_scale (gst_adapter_available (priv->adapter),
           GST_SECOND, ctx->info.rate * ctx->info.bpf);
 
-      if (G_UNLIKELY (shift > GST_BUFFER_TIMESTAMP (buffer))) {
+      if (G_UNLIKELY (shift > GST_BUFFER_PTS (buffer))) {
         /* ERROR */
         goto wrong_time;
       }
       /* arrange for newly added samples to come out with the ts
        * of the incoming buffer that adds these */
-      priv->base_ts = GST_BUFFER_TIMESTAMP (buffer) - shift;
+      priv->base_ts = GST_BUFFER_PTS (buffer) - shift;
       priv->samples = 0;
       gst_audio_encoder_set_base_gp (enc);
       priv->discont |= discont;