mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2024-07-03 05:05:52 +00:00
Code Issues 1 Projects Releases Wiki Activity

gst/audiorate/gstaudiorate.c: Keep sink and src segment to keep track of time and support more input formats.

Browse source 
Original commit message from CVS:
* gst/audiorate/gstaudiorate.c: (gst_audio_rate_reset),
(gst_audio_rate_sink_event), (gst_audio_rate_convert),
(gst_audio_rate_convert_segments), (gst_audio_rate_chain):
Keep sink and src segment to keep track of time and support more
input formats.
Fix bogus next_offset and run_time calculation, don't understand how
this could have worked before. Fixes #357976.
Remove some unneeded vars.
This commit is contained in:
Wim Taymans 2006-09-28 10:49:56 +00:00
parent 0c3733c652
commit e10e9eeff2
2 changed files with 147 additions and 32 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

11
ChangeLog
View file

@ -1,3 +1,14 @@
2006-09-28 Wim Taymans <wim@fluendo.com>
* gst/audiorate/gstaudiorate.c: (gst_audio_rate_reset),
(gst_audio_rate_sink_event), (gst_audio_rate_convert),
(gst_audio_rate_convert_segments), (gst_audio_rate_chain):
Keep sink and src segment to keep track of time and support more
input formats.
Fix bogus next_offset and run_time calculation, don't understand how
this could have worked before. Fixes #357976.
Remove some unneeded vars.
2006-09-28 Tim-Philipp Müller <tim at centricular dot net>
* gst/playback/gstplaybin.c: (remove_sinks):

168
gst/audiorate/gstaudiorate.c
View file

@ -57,11 +57,15 @@ struct _GstAudioRate
gboolean silent;
/* audio state */
guint64 offset;
guint64 next_offset;
gboolean discont;
GstSegment segment;
gboolean new_segment;
/* we accept all formats on the sink */
GstSegment sink_segment;
/* we output TIME format on the src */
GstSegment src_segment;
};
struct _GstAudioRateClass
@ -202,10 +206,10 @@ gst_audio_rate_class_init (GstAudioRateClass * klass)
static void
gst_audio_rate_reset (GstAudioRate * audiorate)
{
audiorate->offset = -1;
audiorate->next_offset = -1;
audiorate->discont = TRUE;
gst_segment_init (&audiorate->segment, GST_FORMAT_UNDEFINED);
gst_segment_init (&audiorate->sink_segment, GST_FORMAT_UNDEFINED);
gst_segment_init (&audiorate->src_segment, GST_FORMAT_TIME);
GST_DEBUG_OBJECT (audiorate, "handle reset");
}
@ -322,17 +326,32 @@ gst_audio_rate_sink_event (GstPad * pad, GstEvent * event)
/* a new segment starts. We need to figure out what will be the next
* sample offset. We mark the offsets as invalid so that the _chain
* function will perform this calculation. */
audiorate->offset = -1;
audiorate->next_offset = -1;
}
gst_segment_set_newsegment_full (&audiorate->segment, update, rate, arate,
format, start, stop, time);
/* we accept all formats */
gst_segment_set_newsegment_full (&audiorate->sink_segment, update, rate,
arate, format, start, stop, time);
res = gst_pad_push_event (audiorate->srcpad, event);
GST_DEBUG_OBJECT (audiorate, "updated segment: %" GST_SEGMENT_FORMAT,
&audiorate->sink_segment);
if (format == GST_FORMAT_TIME) {
/* TIME formats can be copied to src and forwarded */
res = gst_pad_push_event (audiorate->srcpad, event);
memcpy (&audiorate->src_segment, &audiorate->sink_segment,
sizeof (GstSegment));
} else {
/* other formats will be handled in the _chain function */
gst_event_unref (event);
res = TRUE;
}
break;
}
case GST_EVENT_EOS:
/* FIXME, fill last segment */
res = gst_pad_push_event (audiorate->srcpad, event);
break;
default:
res = gst_pad_push_event (audiorate->srcpad, event);
break;
@ -362,56 +381,141 @@ gst_audio_rate_src_event (GstPad * pad, GstEvent * event)
return res;
}
static gboolean
gst_audio_rate_convert (GstAudioRate * audiorate,
GstFormat src_fmt, gint64 src_val, GstFormat dest_fmt, gint64 * dest_val)
{
if (src_fmt == dest_fmt) {
*dest_val = src_val;
return TRUE;
}
switch (src_fmt) {
case GST_FORMAT_DEFAULT:
switch (dest_fmt) {
case GST_FORMAT_BYTES:
*dest_val = src_val * audiorate->bytes_per_sample;
break;
case GST_FORMAT_TIME:
*dest_val =
gst_util_uint64_scale_int (src_val, GST_SECOND, audiorate->rate);
break;
default:
return FALSE;;
}
break;
case GST_FORMAT_BYTES:
switch (dest_fmt) {
case GST_FORMAT_DEFAULT:
*dest_val = src_val / audiorate->bytes_per_sample;
break;
case GST_FORMAT_TIME:
*dest_val = gst_util_uint64_scale_int (src_val, GST_SECOND,
audiorate->rate * audiorate->bytes_per_sample);
break;
default:
return FALSE;;
}
break;
case GST_FORMAT_TIME:
switch (dest_fmt) {
case GST_FORMAT_BYTES:
*dest_val = gst_util_uint64_scale_int (src_val,
audiorate->rate * audiorate->bytes_per_sample, GST_SECOND);
break;
case GST_FORMAT_DEFAULT:
*dest_val =
gst_util_uint64_scale_int (src_val, audiorate->rate, GST_SECOND);
break;
default:
return FALSE;;
}
break;
default:
return FALSE;
}
return TRUE;
}
static gboolean
gst_audio_rate_convert_segments (GstAudioRate * audiorate)
{
GstFormat src_fmt, dst_fmt;
src_fmt = audiorate->sink_segment.format;
dst_fmt = audiorate->src_segment.format;
#define CONVERT_VAL(field) gst_audio_rate_convert (audiorate, \
src_fmt, audiorate->sink_segment.field, \
dst_fmt, &audiorate->src_segment.field);
audiorate->sink_segment.rate = audiorate->src_segment.rate;
audiorate->sink_segment.abs_rate = audiorate->src_segment.abs_rate;
audiorate->sink_segment.flags = audiorate->src_segment.flags;
audiorate->sink_segment.applied_rate = audiorate->src_segment.applied_rate;
CONVERT_VAL (start);
CONVERT_VAL (stop);
CONVERT_VAL (time);
CONVERT_VAL (accum);
CONVERT_VAL (last_stop);
#undef CONVERT_VAL
return TRUE;
}
static GstFlowReturn
gst_audio_rate_chain (GstPad * pad, GstBuffer * buf)
{
GstAudioRate *audiorate;
GstClockTime in_time, in_duration, run_time;
guint64 in_offset, in_offset_end;
GstClockTime in_time, in_duration, in_stop, run_time;
guint64 in_offset, in_offset_end, in_samples;
guint in_size;
GstFlowReturn ret = GST_FLOW_OK;
audiorate = GST_AUDIO_RATE (gst_pad_get_parent (pad));
/* need to be negotiated now */
if (audiorate->bytes_per_sample == 0)
goto not_negotiated;
if (audiorate->offset == -1) {
/* we have a new pending segment */
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.time, which is the first
* 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
* buffer in that segment, which is the offset expressed in DEFAULT units.
*/
pos = audiorate->segment.time;
if (pos != 0) {
if (audiorate->segment.format == GST_FORMAT_TIME) {
/* convert first timestamp of segment to sample position */
pos = gst_util_uint64_scale_int (pos, audiorate->rate, GST_SECOND);
} else {
/* FIXME, we don't know, start from 0 then... */
pos = 0;
}
}
/* convert first timestamp of segment to sample position */
pos = gst_util_uint64_scale_int (audiorate->src_segment.start,
audiorate->rate, GST_SECOND);
GST_DEBUG_OBJECT (audiorate, "resync to offset %" G_GINT64_FORMAT, pos);
audiorate->offset = pos;
audiorate->next_offset = pos;
}
audiorate->in++;
in_time = GST_BUFFER_TIMESTAMP (buf);
in_duration = GST_BUFFER_DURATION (buf);
in_size = GST_BUFFER_SIZE (buf);
in_samples = in_size / audiorate->bytes_per_sample;
/* get duration from the size because we can and it's more accurate */
in_duration =
gst_util_uint64_scale_int (in_samples, GST_SECOND, audiorate->rate);
in_stop = in_time + in_duration;
/* don't really on buffer's offset */
/* We instead figure out using the runningtime version of the incoming buffer timestamp */
run_time =
gst_segment_to_running_time (&audiorate->segment, GST_FORMAT_TIME,
in_time);
/* Figure out the total accumulated segment time. */
run_time = in_time + audiorate->src_segment.accum;
/* calculate the buffer offset */
in_offset = gst_util_uint64_scale_int (run_time, audiorate->rate, GST_SECOND);
in_offset_end = in_offset + in_size / audiorate->bytes_per_sample;
in_offset_end = in_offset + in_samples;
GST_LOG_OBJECT (audiorate,
"in_time:%" GST_TIME_FORMAT ", run_time:%" GST_TIME_FORMAT
@ -431,6 +535,7 @@ gst_audio_rate_chain (GstPad * pad, GstBuffer * buf)
fillsize = fillsamples * audiorate->bytes_per_sample;
fill = gst_buffer_new_and_alloc (fillsize);
/* FIXME, 0 might not be the silence byte for the negotiated format. */
memset (GST_BUFFER_DATA (fill), 0, fillsize);
GST_LOG_OBJECT (audiorate, "inserting %lld samples", fillsamples);
@ -513,7 +618,7 @@ gst_audio_rate_chain (GstPad * pad, GstBuffer * buf)
}
/* set last_stop on segment */
gst_segment_set_last_stop (&audiorate->segment, GST_FORMAT_TIME,
gst_segment_set_last_stop (&audiorate->src_segment, GST_FORMAT_TIME,
GST_BUFFER_TIMESTAMP (buf) + GST_BUFFER_DURATION (buf));
ret = gst_pad_push (audiorate->srcpad, buf);
@ -521,7 +626,6 @@ gst_audio_rate_chain (GstPad * pad, GstBuffer * buf)
audiorate->next_offset = in_offset_end;
beach:
audiorate->offset += in_size / audiorate->bytes_per_sample;
gst_object_unref (audiorate);