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