audiobuffersplit: Keep incoming and outgoing segments separate

We might have to drain already queued input based on the old segment
before forwarding the new segment event. The new segment is only
forwarded after a discont as otherwise we might cause unnecessary
timestamp jumps as we output buffers timestamped based on sample counts.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1602>

2 files changed, 35 insertions, 11 deletions

diff --git a/gst/audiobuffersplit/gstaudiobuffersplit.c b/gst/audiobuffersplit/gstaudiobuffersplit.c
index d27dcf7e5..08514cdb8 100644
--- a/gst/audiobuffersplit/gstaudiobuffersplit.c
+++ b/gst/audiobuffersplit/gstaudiobuffersplit.c
@@ -321,7 +321,9 @@ gst_audio_buffer_split_change_state (GstElement * element,
   switch (transition) {
     case GST_STATE_CHANGE_READY_TO_PAUSED:
       gst_audio_info_init (&self->info);
-      gst_segment_init (&self->segment, GST_FORMAT_TIME);
+      gst_segment_init (&self->in_segment, GST_FORMAT_TIME);
+      gst_segment_init (&self->out_segment, GST_FORMAT_UNDEFINED);
+      self->segment_pending = FALSE;
       GST_OBJECT_LOCK (self);
       gst_audio_stream_align_mark_discont (self->stream_align);
       GST_OBJECT_UNLOCK (self);
@@ -389,7 +391,7 @@ gst_audio_buffer_split_output (GstAudioBufferSplit * self, gboolean force,
 
     resync_time_diff =
         gst_util_uint64_scale (self->current_offset, GST_SECOND, rate);
-    if (self->segment.rate < 0.0) {
+    if (self->out_segment.rate < 0.0) {
       if (resync_time > resync_time_diff)
         GST_BUFFER_TIMESTAMP (buffer) = resync_time - resync_time_diff;
       else
@@ -451,7 +453,7 @@ gst_audio_buffer_split_handle_discont (GstAudioBufferSplit * self,
   GST_OBJECT_LOCK (self);
   discont =
       gst_audio_stream_align_process (self->stream_align,
-      self->segment.rate < 0 ? FALSE : GST_BUFFER_IS_DISCONT (buffer)
+      self->in_segment.rate < 0 ? FALSE : GST_BUFFER_IS_DISCONT (buffer)
       || GST_BUFFER_FLAG_IS_SET (buffer, GST_BUFFER_FLAG_RESYNC),
       GST_BUFFER_PTS (buffer), gst_buffer_get_size (buffer) / bpf, NULL, NULL,
       NULL);
@@ -463,7 +465,7 @@ gst_audio_buffer_split_handle_discont (GstAudioBufferSplit * self,
   /* Reset */
   self->drop_samples = 0;
 
-  if (self->segment.rate < 0.0) {
+  if (self->in_segment.rate < 0.0) {
     current_timestamp =
         self->resync_time - gst_util_uint64_scale (self->current_offset +
         avail_samples, GST_SECOND, rate);
@@ -589,6 +591,18 @@ gst_audio_buffer_split_handle_discont (GstAudioBufferSplit * self,
     self->current_offset = 0;
     self->accumulated_error = 0;
     self->resync_time = GST_BUFFER_PTS (buffer);
+
+    if (self->segment_pending) {
+      GstEvent *event;
+
+      self->out_segment = self->in_segment;
+      GST_DEBUG_OBJECT (self, "Updating output segment %" GST_SEGMENT_FORMAT,
+          &self->out_segment);
+      event = gst_event_new_segment (&self->out_segment);
+      gst_event_set_seqnum (event, self->segment_seqnum);
+      gst_pad_push_event (self->srcpad, event);
+      self->segment_pending = FALSE;
+    }
   }
 
   return ret;
@@ -621,7 +635,7 @@ gst_audio_buffer_split_clip_buffer_start_for_gapless (GstAudioBufferSplit *
     return NULL;
   }
 
-  if (self->segment.rate < 0.0) {
+  if (self->out_segment.rate < 0.0) {
     buffer =
         gst_audio_buffer_truncate (buffer, bpf, 0,
         nsamples - self->drop_samples);
@@ -660,7 +674,7 @@ gst_audio_buffer_split_sink_chain (GstPad * pad, GstObject * parent,
   }
 
   buffer =
-      gst_audio_buffer_split_clip_buffer (self, buffer, &self->segment, rate,
+      gst_audio_buffer_split_clip_buffer (self, buffer, &self->in_segment, rate,
       bpf);
   if (!buffer)
     return GST_FLOW_OK;
@@ -741,7 +755,9 @@ gst_audio_buffer_split_sink_event (GstPad * pad, GstObject * parent,
       break;
     }
     case GST_EVENT_FLUSH_STOP:
-      gst_segment_init (&self->segment, GST_FORMAT_TIME);
+      gst_segment_init (&self->in_segment, GST_FORMAT_TIME);
+      gst_segment_init (&self->out_segment, GST_FORMAT_UNDEFINED);
+      self->segment_pending = FALSE;
       GST_OBJECT_LOCK (self);
       gst_audio_stream_align_mark_discont (self->stream_align);
       GST_OBJECT_UNLOCK (self);
@@ -751,12 +767,18 @@ gst_audio_buffer_split_sink_event (GstPad * pad, GstObject * parent,
       ret = gst_pad_event_default (pad, parent, event);
       break;
     case GST_EVENT_SEGMENT:
-      gst_event_copy_segment (event, &self->segment);
-      if (self->segment.format != GST_FORMAT_TIME) {
+      gst_event_copy_segment (event, &self->in_segment);
+      if (self->in_segment.format != GST_FORMAT_TIME) {
         gst_event_unref (event);
         ret = FALSE;
       } else {
-        ret = gst_pad_event_default (pad, parent, event);
+        GST_DEBUG_OBJECT (self,
+            "Received new input segment %" GST_SEGMENT_FORMAT,
+            &self->in_segment);
+        self->segment_pending = TRUE;
+        self->segment_seqnum = gst_event_get_seqnum (event);
+        gst_event_unref (event);
+        ret = TRUE;
       }
       break;
     case GST_EVENT_EOS:
diff --git a/gst/audiobuffersplit/gstaudiobuffersplit.h b/gst/audiobuffersplit/gstaudiobuffersplit.h
index 8625df8ca..d22eba682 100644
--- a/gst/audiobuffersplit/gstaudiobuffersplit.h
+++ b/gst/audiobuffersplit/gstaudiobuffersplit.h
@@ -47,7 +47,9 @@ struct _GstAudioBufferSplit {
   gint output_buffer_duration_d;
 
   /* State */
-  GstSegment segment;
+  GstSegment in_segment, out_segment;
+  guint32 segment_seqnum;
+  gboolean segment_pending;
   GstAudioInfo info;
 
   GstAdapter *adapter;