mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2024-11-23 02:01:12 +00:00
Code Issues 1 Projects Releases Wiki Activity

ext/vorbis/vorbisenc.*: Keep track of the upstream segments and use the running time on that segment instead of the b...

Browse source 
Original commit message from CVS:
Based on a patch by: Klaas <klaas at rivercrew dot net>
* ext/vorbis/vorbisenc.c: (gst_vorbis_enc_sink_event),
(gst_vorbis_enc_buffer_check_discontinuous),
(gst_vorbis_enc_chain), (gst_vorbis_enc_change_state):
* ext/vorbis/vorbisenc.h:
Keep track of the upstream segments and use the running time on that
segment instead of the buffer timestamp everywhere. Fixes bug #525807.
This commit is contained in:
Klaas 2008-10-08 12:08:01 +00:00 committed by Sebastian Dröge
parent c915582c17
commit 93f2eaa98c
3 changed files with 53 additions and 25 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

11
ChangeLog
View file

@ -1,3 +1,14 @@
2008-10-08 Sebastian Dröge <sebastian.droege@collabora.co.uk>
Based on a patch by: Klaas <klaas at rivercrew dot net>
* ext/vorbis/vorbisenc.c: (gst_vorbis_enc_sink_event),
(gst_vorbis_enc_buffer_check_discontinuous),
(gst_vorbis_enc_chain), (gst_vorbis_enc_change_state):
* ext/vorbis/vorbisenc.h:
Keep track of the upstream segments and use the running time on that
segment instead of the buffer timestamp everywhere. Fixes bug #525807.
2008-10-08 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/audioconvert/audioconvert.c: (audio_convert_convert):

66
ext/vorbis/vorbisenc.c
View file

@ -1033,6 +1033,20 @@ gst_vorbis_enc_sink_event (GstPad * pad, GstEvent * event)
}
res = gst_pad_push_event (vorbisenc->srcpad, event);
break;
case GST_EVENT_NEWSEGMENT:
{
gboolean update;
gdouble rate, applied_rate;
GstFormat format;
gint64 start, stop, position;
gst_event_parse_new_segment_full (event, &update, &rate, &applied_rate,
&format, &start, &stop, &position);
if (format == GST_FORMAT_TIME)
gst_segment_set_newsegment (&vorbisenc->segment, update, rate, format,
start, stop, position);
}
/* fall through */
default:
res = gst_pad_push_event (vorbisenc->srcpad, event);
break;
@ -1042,33 +1056,29 @@ gst_vorbis_enc_sink_event (GstPad * pad, GstEvent * event)
static gboolean
gst_vorbis_enc_buffer_check_discontinuous (GstVorbisEnc * vorbisenc,
GstBuffer * buffer)
GstClockTime timestamp, GstClockTime duration)
{
gboolean ret = FALSE;
if (GST_BUFFER_TIMESTAMP (buffer) != GST_CLOCK_TIME_NONE &&
if (timestamp != GST_CLOCK_TIME_NONE &&
vorbisenc->expected_ts != GST_CLOCK_TIME_NONE &&
GST_BUFFER_TIMESTAMP (buffer) != vorbisenc->expected_ts) {
duration != vorbisenc->expected_ts) {
/* It turns out that a lot of elements don't generate perfect streams due
* to rounding errors. So, we permit small errors (< 1/2 a sample) without
* causing a discont.
*/
int halfsample = GST_SECOND / vorbisenc->frequency / 2;
if ((GstClockTimeDiff) (GST_BUFFER_TIMESTAMP (buffer) -
vorbisenc->expected_ts) > halfsample) {
if ((GstClockTimeDiff) (timestamp - vorbisenc->expected_ts) > halfsample) {
GST_DEBUG_OBJECT (vorbisenc, "Expected TS %" GST_TIME_FORMAT
", buffer TS %" GST_TIME_FORMAT,
GST_TIME_ARGS (vorbisenc->expected_ts),
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buffer)));
GST_TIME_ARGS (vorbisenc->expected_ts), GST_TIME_ARGS (timestamp));
ret = TRUE;
}
}
if (GST_BUFFER_TIMESTAMP (buffer) != GST_CLOCK_TIME_NONE &&
GST_BUFFER_DURATION (buffer) != GST_CLOCK_TIME_NONE) {
vorbisenc->expected_ts = GST_BUFFER_TIMESTAMP (buffer) +
GST_BUFFER_DURATION (buffer);
if (timestamp != GST_CLOCK_TIME_NONE && duration != GST_CLOCK_TIME_NONE) {
vorbisenc->expected_ts = timestamp + duration;
} else
vorbisenc->expected_ts = GST_CLOCK_TIME_NONE;
@ -1086,12 +1096,17 @@ gst_vorbis_enc_chain (GstPad * pad, GstBuffer * buffer)
float **vorbis_buffer;
GstBuffer *buf1, *buf2, *buf3;
gboolean first = FALSE;
GstClockTime timestamp = GST_CLOCK_TIME_NONE;
vorbisenc = GST_VORBISENC (GST_PAD_PARENT (pad));
if (!vorbisenc->setup)
goto not_setup;
timestamp =
gst_segment_to_running_time (&vorbisenc->segment, GST_FORMAT_TIME,
GST_BUFFER_TIMESTAMP (buffer));
if (!vorbisenc->header_sent) {
/* Vorbis streams begin with three headers; the initial header (with
most of the codec setup parameters) which is mandated by the Ogg
@ -1148,10 +1163,10 @@ gst_vorbis_enc_chain (GstPad * pad, GstBuffer * buffer)
/* now adjust starting granulepos accordingly if the buffer's timestamp is
nonzero */
vorbisenc->next_ts = GST_BUFFER_TIMESTAMP (buffer);
vorbisenc->expected_ts = GST_BUFFER_TIMESTAMP (buffer);
vorbisenc->next_ts = timestamp;
vorbisenc->expected_ts = timestamp;
vorbisenc->granulepos_offset = gst_util_uint64_scale
(GST_BUFFER_TIMESTAMP (buffer), vorbisenc->frequency, GST_SECOND);
(timestamp, vorbisenc->frequency, GST_SECOND);
vorbisenc->subgranule_offset = 0;
vorbisenc->subgranule_offset =
vorbisenc->next_ts - granulepos_to_timestamp_offset (vorbisenc, 0);
@ -1161,15 +1176,14 @@ gst_vorbis_enc_chain (GstPad * pad, GstBuffer * buffer)
}
if (vorbisenc->expected_ts != GST_CLOCK_TIME_NONE &&
GST_BUFFER_TIMESTAMP (buffer) < vorbisenc->expected_ts) {
guint64 diff = vorbisenc->expected_ts - GST_BUFFER_TIMESTAMP (buffer);
timestamp < vorbisenc->expected_ts) {
guint64 diff = vorbisenc->expected_ts - timestamp;
guint64 diff_bytes;
GST_WARNING_OBJECT (vorbisenc, "Buffer is older than previous "
"timestamp + duration (%" GST_TIME_FORMAT "< %" GST_TIME_FORMAT
"), cannot handle. Clipping buffer.",
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buffer)),
GST_TIME_ARGS (vorbisenc->expected_ts));
GST_TIME_ARGS (timestamp), GST_TIME_ARGS (vorbisenc->expected_ts));
diff_bytes =
GST_CLOCK_TIME_TO_FRAMES (diff,
@ -1187,11 +1201,12 @@ gst_vorbis_enc_chain (GstPad * pad, GstBuffer * buffer)
GST_BUFFER_DURATION (buffer) -= diff;
}
if (gst_vorbis_enc_buffer_check_discontinuous (vorbisenc, buffer) && !first) {
GST_WARNING_OBJECT (vorbisenc, "Buffer is discontinuous, flushing encoder "
"and restarting (Discont from %" GST_TIME_FORMAT
" to %" GST_TIME_FORMAT ")", GST_TIME_ARGS (vorbisenc->next_ts),
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buffer)));
if (gst_vorbis_enc_buffer_check_discontinuous (vorbisenc, timestamp,
GST_BUFFER_DURATION (buffer)) && !first) {
GST_WARNING_OBJECT (vorbisenc,
"Buffer is discontinuous, flushing encoder "
"and restarting (Discont from %" GST_TIME_FORMAT " to %" GST_TIME_FORMAT
")", GST_TIME_ARGS (vorbisenc->next_ts), GST_TIME_ARGS (timestamp));
/* Re-initialise encoder (there's unfortunately no API to flush it) */
if ((ret = gst_vorbis_enc_clear (vorbisenc)) != GST_FLOW_OK)
return ret;
@ -1200,11 +1215,11 @@ gst_vorbis_enc_chain (GstPad * pad, GstBuffer * buffer)
we successfully initialised earlier */
/* Now, set our granulepos offset appropriately. */
vorbisenc->next_ts = GST_BUFFER_TIMESTAMP (buffer);
vorbisenc->next_ts = timestamp;
/* We need to round to the nearest whole number of samples, not just do
* a truncating division here */
vorbisenc->granulepos_offset = gst_util_uint64_scale
(GST_BUFFER_TIMESTAMP (buffer) + GST_SECOND / vorbisenc->frequency / 2
(timestamp + GST_SECOND / vorbisenc->frequency / 2
- vorbisenc->subgranule_offset, vorbisenc->frequency, GST_SECOND);
vorbisenc->header_sent = TRUE;
@ -1418,6 +1433,7 @@ gst_vorbis_enc_change_state (GstElement * element, GstStateChange transition)
vorbisenc->setup = FALSE;
vorbisenc->next_discont = FALSE;
vorbisenc->header_sent = FALSE;
gst_segment_init (&vorbisenc->segment, GST_FORMAT_TIME);
break;
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
break;

1
ext/vorbis/vorbisenc.h
View file

@ -80,6 +80,7 @@ struct _GstVorbisEnc {
gboolean next_discont;
guint64 granulepos_offset;
gint64 subgranule_offset;
GstSegment segment;
GstTagList * tags;