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Revert "audiobuffersplit: Update out_segment even without discont"
This reverts commit c0dc65d40a
.
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parent
9426eaae6a
commit
28b64bc28a
1 changed files with 38 additions and 34 deletions
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@ -564,7 +564,7 @@ gst_audio_buffer_split_handle_discont (GstAudioBufferSplit * self,
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drop_samples, GST_TIME_ARGS (gst_util_uint64_scale (drop_samples,
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GST_SECOND, rate)));
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self->drop_samples = drop_samples;
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return ret;
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discont = FALSE;
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} else if (new_offset > self->current_offset + avail_samples) {
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guint64 silence_samples =
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new_offset - (self->current_offset + avail_samples);
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@ -604,7 +604,7 @@ gst_audio_buffer_split_handle_discont (GstAudioBufferSplit * self,
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silence_samples -= n_samples;
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}
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return ret;
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discont = FALSE;
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}
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} else if (new_offset < self->current_offset + avail_samples) {
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guint64 drop_samples = self->current_offset + avail_samples - new_offset;
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@ -614,32 +614,48 @@ gst_audio_buffer_split_handle_discont (GstAudioBufferSplit * self,
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drop_samples, GST_TIME_ARGS (gst_util_uint64_scale (drop_samples,
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GST_SECOND, rate)));
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self->drop_samples = drop_samples;
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return ret;
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discont = FALSE;
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}
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}
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/* We might end up in here also in gapless mode, if the above code decided
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* that no silence is to be inserted, because e.g. the gap is too big */
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GST_DEBUG_OBJECT (self, "Got %s: Current running time %" GST_TIME_FORMAT
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", current end running time %" GST_TIME_FORMAT
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", running time after discont %" GST_TIME_FORMAT,
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self->current_offset == -1 ? "first buffer" : "discont",
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GST_TIME_ARGS (current_rt), GST_TIME_ARGS (current_rt_end),
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GST_TIME_ARGS (input_rt));
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if (discont) {
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/* We might end up in here also in gapless mode, if the above code decided
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* that no silence is to be inserted, because e.g. the gap is too big */
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GST_DEBUG_OBJECT (self,
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"Got %s: Current running time %" GST_TIME_FORMAT
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", current end running time %" GST_TIME_FORMAT
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", running time after discont %" GST_TIME_FORMAT,
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self->current_offset == -1 ? "first buffer" : "discont",
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GST_TIME_ARGS (current_rt),
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GST_TIME_ARGS (current_rt_end), GST_TIME_ARGS (input_rt));
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if (self->strict_buffer_size) {
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gst_adapter_clear (self->adapter);
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ret = GST_FLOW_OK;
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} else {
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ret = gst_audio_buffer_split_output (self, TRUE, rate, bpf,
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samples_per_buffer);
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if (self->strict_buffer_size) {
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gst_adapter_clear (self->adapter);
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ret = GST_FLOW_OK;
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} else {
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ret =
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gst_audio_buffer_split_output (self, TRUE, rate, bpf,
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samples_per_buffer);
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}
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self->current_offset = 0;
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self->accumulated_error = 0;
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self->resync_pts = GST_BUFFER_PTS (buffer);
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self->resync_rt = input_rt;
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if (self->segment_pending) {
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GstEvent *event;
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self->out_segment = self->in_segment;
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GST_DEBUG_OBJECT (self, "Updating output segment %" GST_SEGMENT_FORMAT,
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&self->out_segment);
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event = gst_event_new_segment (&self->out_segment);
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gst_event_set_seqnum (event, self->segment_seqnum);
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gst_pad_push_event (self->srcpad, event);
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self->segment_pending = FALSE;
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}
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}
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self->current_offset = 0;
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self->accumulated_error = 0;
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self->resync_pts = GST_BUFFER_PTS (buffer);
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self->resync_rt = input_rt;
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return ret;
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}
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@ -723,18 +739,6 @@ gst_audio_buffer_split_sink_chain (GstPad * pad, GstObject * parent,
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return ret;
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}
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if (self->segment_pending) {
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GstEvent *event;
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self->out_segment = self->in_segment;
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GST_DEBUG_OBJECT (self, "Updating output segment %" GST_SEGMENT_FORMAT,
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&self->out_segment);
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event = gst_event_new_segment (&self->out_segment);
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gst_event_set_seqnum (event, self->segment_seqnum);
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gst_pad_push_event (self->srcpad, event);
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self->segment_pending = FALSE;
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}
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buffer =
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gst_audio_buffer_split_clip_buffer_start_for_gapless (self, buffer, rate,
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bpf);
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