mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-11-23 10:11:08 +00:00
ext/vorbis/vorbisenc.*: Keep track of the upstream segments and use the running time on that segment instead of the b...
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:
parent
c915582c17
commit
93f2eaa98c
3 changed files with 53 additions and 25 deletions
11
ChangeLog
11
ChangeLog
|
@ -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):
|
||||
|
|
|
@ -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;
|
||||
|
|
|
@ -80,6 +80,7 @@ struct _GstVorbisEnc {
|
|||
gboolean next_discont;
|
||||
guint64 granulepos_offset;
|
||||
gint64 subgranule_offset;
|
||||
GstSegment segment;
|
||||
|
||||
GstTagList * tags;
|
||||
|
||||
|
|
Loading…
Reference in a new issue