mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-11-27 12:11:13 +00:00
gst/audiorate/gstaudiorate.c: Fix audiorate, so that it accurately sets offsets and timestamps.
Original commit message from CVS: * gst/audiorate/gstaudiorate.c: (gst_audio_rate_reset), (gst_audio_rate_sink_event), (gst_audio_rate_chain): Fix audiorate, so that it accurately sets offsets and timestamps. Doesn't change the fundamental algorithmic decisions; so should be safe. * tests/check/Makefile.am: Enable audiorate test now that it passes.
This commit is contained in:
Michael Smith
parent
8795eafaa4
commit
4ac9b64fd6
3 changed files with 52 additions and 14 deletions
11
ChangeLog
11
ChangeLog
|
|
@ -1,3 +1,14 @@
|
|||
2006-11-16 Michael Smith <msmith@fluendo.com>
|
||||
|
||||
* gst/audiorate/gstaudiorate.c: (gst_audio_rate_reset),
|
||||
(gst_audio_rate_sink_event), (gst_audio_rate_chain):
|
||||
Fix audiorate, so that it accurately sets offsets and timestamps.
|
||||
Doesn't change the fundamental algorithmic decisions; so should be
|
||||
safe.
|
||||
|
||||
* tests/check/Makefile.am:
|
||||
Enable audiorate test now that it passes.
|
||||
|
||||
2006-11-09 Stefan Kost <ensonic@users.sf.net>
|
||||
|
||||
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_change_state):
|
||||
|
|
|
|||
|
|
@ -17,6 +17,8 @@
|
|||
* Boston, MA 02111-1307, USA.
|
||||
*/
|
||||
|
||||
#include <stdlib.h>
|
||||
|
||||
#ifdef HAVE_CONFIG_H
|
||||
#include "config.h"
|
||||
#endif
|
||||
|
|
@ -58,6 +60,7 @@ struct _GstAudioRate
|
|||
|
||||
/* audio state */
|
||||
guint64 next_offset;
|
||||
guint64 next_ts;
|
||||
|
||||
gboolean discont;
|
||||
|
||||
|
|
@ -207,6 +210,7 @@ static void
|
|||
gst_audio_rate_reset (GstAudioRate * audiorate)
|
||||
{
|
||||
audiorate->next_offset = -1;
|
||||
audiorate->next_ts = -1;
|
||||
audiorate->discont = TRUE;
|
||||
gst_segment_init (&audiorate->sink_segment, GST_FORMAT_UNDEFINED);
|
||||
gst_segment_init (&audiorate->src_segment, GST_FORMAT_TIME);
|
||||
|
|
@ -327,6 +331,7 @@ gst_audio_rate_sink_event (GstPad * pad, GstEvent * event)
|
|||
* sample offset. We mark the offsets as invalid so that the _chain
|
||||
* function will perform this calculation. */
|
||||
audiorate->next_offset = -1;
|
||||
audiorate->next_ts = -1;
|
||||
}
|
||||
|
||||
/* we accept all formats */
|
||||
|
|
@ -498,10 +503,21 @@ gst_audio_rate_chain (GstPad * pad, GstBuffer * buf)
|
|||
GST_DEBUG_OBJECT (audiorate, "resync to offset %" G_GINT64_FORMAT, pos);
|
||||
|
||||
audiorate->next_offset = pos;
|
||||
audiorate->next_ts = gst_util_uint64_scale_int (audiorate->next_offset,
|
||||
GST_SECOND, audiorate->rate);
|
||||
}
|
||||
|
||||
audiorate->in++;
|
||||
|
||||
static guint64 nextts = 0;
|
||||
|
||||
#define LLABS(a) ((gint64)(a) < 0 ? -(a):(a))
|
||||
if (nextts != GST_BUFFER_TIMESTAMP (buf) && LLABS (GST_BUFFER_TIMESTAMP (buf) - nextts) > 21000) /* 21 us, ~1 sample */
|
||||
GST_DEBUG_OBJECT (audiorate, "Expected %lld, got %lld! --> %lld", nextts,
|
||||
GST_BUFFER_TIMESTAMP (buf),
|
||||
LLABS (GST_BUFFER_TIMESTAMP (buf) - nextts));
|
||||
nextts = GST_BUFFER_TIMESTAMP (buf) + GST_BUFFER_DURATION (buf);
|
||||
|
||||
in_time = GST_BUFFER_TIMESTAMP (buf);
|
||||
in_size = GST_BUFFER_SIZE (buf);
|
||||
in_samples = in_size / audiorate->bytes_per_sample;
|
||||
|
|
@ -538,14 +554,21 @@ gst_audio_rate_chain (GstPad * pad, GstBuffer * buf)
|
|||
/* 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);
|
||||
GST_DEBUG_OBJECT (audiorate, "inserting %lld samples", fillsamples);
|
||||
|
||||
GST_BUFFER_DURATION (fill) = in_duration * fillsize / in_size;
|
||||
GST_BUFFER_TIMESTAMP (fill) = in_time - GST_BUFFER_DURATION (fill);
|
||||
GST_BUFFER_OFFSET (fill) = audiorate->next_offset;
|
||||
GST_BUFFER_OFFSET_END (fill) = in_offset;
|
||||
audiorate->next_offset += fillsamples;
|
||||
GST_BUFFER_OFFSET_END (fill) = audiorate->next_offset;
|
||||
|
||||
/* we created this buffer to filla gap */
|
||||
/* Use next timestamp, then calculate following timestamp based on in_offset
|
||||
* to get duration. Neccesary complexity to get 'perfect' streams */
|
||||
GST_BUFFER_TIMESTAMP (fill) = audiorate->next_ts;
|
||||
audiorate->next_ts = gst_util_uint64_scale_int (in_offset,
|
||||
GST_SECOND, audiorate->rate);
|
||||
GST_BUFFER_DURATION (fill) = audiorate->next_ts -
|
||||
GST_BUFFER_TIMESTAMP (fill);
|
||||
|
||||
/* we created this buffer to fill a gap */
|
||||
GST_BUFFER_FLAG_SET (fill, GST_BUFFER_FLAG_GAP);
|
||||
/* set discont if it's pending, this is mostly done for the first buffer and
|
||||
* after a flushing seek */
|
||||
|
|
@ -570,7 +593,7 @@ gst_audio_rate_chain (GstPad * pad, GstBuffer * buf)
|
|||
|
||||
audiorate->drop += drop;
|
||||
|
||||
GST_LOG_OBJECT (audiorate, "dropping %lld samples", drop);
|
||||
GST_DEBUG_OBJECT (audiorate, "dropping %lld samples", drop);
|
||||
|
||||
/* we can drop the buffer completely */
|
||||
gst_buffer_unref (buf);
|
||||
|
|
@ -590,13 +613,6 @@ gst_audio_rate_chain (GstPad * pad, GstBuffer * buf)
|
|||
leftsize = in_size - truncsize;
|
||||
|
||||
trunc = gst_buffer_create_sub (buf, truncsize, leftsize);
|
||||
GST_BUFFER_DURATION (trunc) = in_duration * leftsize / in_size;
|
||||
GST_BUFFER_TIMESTAMP (trunc) =
|
||||
in_time + in_duration - GST_BUFFER_DURATION (trunc);
|
||||
GST_BUFFER_OFFSET (trunc) = audiorate->next_offset;
|
||||
GST_BUFFER_OFFSET_END (trunc) = in_offset_end;
|
||||
|
||||
GST_LOG_OBJECT (audiorate, "truncating %lld samples", truncsamples);
|
||||
|
||||
gst_buffer_unref (buf);
|
||||
buf = trunc;
|
||||
|
|
@ -606,6 +622,17 @@ gst_audio_rate_chain (GstPad * pad, GstBuffer * buf)
|
|||
audiorate->drop += truncsamples;
|
||||
}
|
||||
}
|
||||
|
||||
/* Now calculate parameters for whichever buffer (either the original
|
||||
* or truncated one) we're pushing. */
|
||||
GST_BUFFER_OFFSET (buf) = audiorate->next_offset;
|
||||
GST_BUFFER_OFFSET_END (buf) = in_offset_end;
|
||||
|
||||
GST_BUFFER_TIMESTAMP (buf) = audiorate->next_ts;
|
||||
audiorate->next_ts = gst_util_uint64_scale_int (in_offset_end,
|
||||
GST_SECOND, audiorate->rate);
|
||||
GST_BUFFER_DURATION (buf) = audiorate->next_ts - GST_BUFFER_TIMESTAMP (buf);
|
||||
|
||||
if (audiorate->discont) {
|
||||
/* we need to output a discont buffer, do so now */
|
||||
GST_DEBUG_OBJECT (audiorate, "marking DISCONT on output buffer");
|
||||
|
|
|
|||
|
|
@ -52,6 +52,7 @@ check_PROGRAMS = \
|
|||
$(check_theora) \
|
||||
elements/adder \
|
||||
elements/audioconvert \
|
||||
elements/audiorate \
|
||||
elements/audioresample \
|
||||
elements/audiotestsrc \
|
||||
elements/gdpdepay \
|
||||
|
|
@ -81,7 +82,6 @@ VALGRIND_TO_FIX = \
|
|||
|
||||
# these tests don't even pass
|
||||
noinst_PROGRAMS = \
|
||||
elements/audiorate \
|
||||
elements/ffmpegcolorspace
|
||||
|
||||
AM_CFLAGS = $(GST_CFLAGS) $(GST_CHECK_CFLAGS)
|
||||
|
|
|
|||
Loading…
Reference in a new issue