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audiorate: Always push updated segment

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Convert segment to TIME format immediately instead of waiting for
_chain() to be called. This fixes converted segment never being pushed
downstream.

Fix the convert function that was copying some fields in the wrong
direction. Add fast copy if segment is already in TIME format.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7977>
This commit is contained in:
Xavier Claessens 2024-11-27 09:26:15 -05:00 committed by GStreamer Marge Bot
parent fbad7b593d
commit 0d8bdaaf17
1 changed files with 18 additions and 19 deletions
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37
subprojects/gst-plugins-base/gst/audiorate/gstaudiorate.c
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@ -130,6 +130,7 @@ static void gst_audio_rate_get_property (GObject * object,
static GstStateChangeReturn gst_audio_rate_change_state (GstElement * element,
GstStateChange transition);
static gboolean gst_audio_rate_convert_segments (GstAudioRate * audiorate);
/*static guint gst_audio_rate_signals[LAST_SIGNAL] = { 0 }; */
@ -341,16 +342,14 @@ gst_audio_rate_sink_event (GstPad * pad, GstObject * parent, GstEvent * event)
GST_DEBUG_OBJECT (audiorate, "updated segment: %" GST_SEGMENT_FORMAT,
&audiorate->sink_segment);
if (audiorate->sink_segment.format == GST_FORMAT_TIME) {
/* TIME formats can be copied to src and forwarded */
res = gst_pad_push_event (audiorate->srcpad, event);
gst_segment_copy_into (&audiorate->sink_segment,
&audiorate->src_segment);
} else {
/* other formats will be handled in the _chain function */
gst_event_unref (event);
res = TRUE;
}
/* Copy sink_segment into src_segment and convert to TIME format. */
gst_audio_rate_convert_segments (audiorate);
/* Push updated segment */
guint32 seqnum = gst_event_get_seqnum (event);
gst_event_take (&event, gst_event_new_segment (&audiorate->src_segment));
gst_event_set_seqnum (event, seqnum);
res = gst_pad_push_event (audiorate->srcpad, event);
break;
}
case GST_EVENT_EOS:
@ -414,14 +413,17 @@ gst_audio_rate_convert_segments (GstAudioRate * audiorate)
src_fmt = audiorate->sink_segment.format;
dst_fmt = audiorate->src_segment.format;
if (src_fmt == dst_fmt) {
gst_segment_copy_into (&audiorate->sink_segment, &audiorate->src_segment);
return TRUE;
}
#define CONVERT_VAL(field) gst_audio_rate_convert (audiorate, \
src_fmt, audiorate->sink_segment.field, \
dst_fmt, &audiorate->src_segment.field);
src_fmt, audiorate->sink_segment.field, \
dst_fmt, &audiorate->src_segment.field);
audiorate->sink_segment.rate = audiorate->src_segment.rate;
audiorate->sink_segment.flags = audiorate->src_segment.flags;
audiorate->sink_segment.applied_rate = audiorate->src_segment.applied_rate;
audiorate->src_segment.rate = audiorate->sink_segment.rate;
audiorate->src_segment.flags = audiorate->sink_segment.flags;
audiorate->src_segment.applied_rate = audiorate->sink_segment.applied_rate;
CONVERT_VAL (start);
CONVERT_VAL (stop);
CONVERT_VAL (time);
@ -469,9 +471,6 @@ gst_audio_rate_chain (GstPad * pad, GstObject * parent, GstBuffer * buf)
if (audiorate->next_offset == -1) {
gint64 pos;
/* update the TIME segment */
gst_audio_rate_convert_segments (audiorate);
/* first buffer, we are negotiated and we have a segment, calculate the
* current expected offsets based on the segment.start, which is the first
* media time of the segment and should match the media time of the first