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audioresample: initial filter transient discarded; unit tests passing

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This commit is contained in:
Leo Singer 2010-12-16 20:30:24 -08:00 committed by Sebastian Dröge
parent b4cd3329a9
commit 87f2422737
1 changed files with 17 additions and 18 deletions
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35
gst/audioresample/gstaudioresample.c
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@ -389,6 +389,8 @@ gst_audio_resample_init_state (GstAudioResample * resample, gint width,
return NULL;
}
funcs->skip_zeros (ret);
return ret;
}
@ -941,9 +943,8 @@ gst_audio_resample_event (GstBaseTransform * base, GstEvent * event)
case GST_EVENT_FLUSH_STOP:
gst_audio_resample_reset_state (resample);
if (resample->state)
resample->count_gap = resample->funcs->get_filt_len (resample->state);
else
resample->count_gap = 0;
resample->funcs->skip_zeros (resample->state);
resample->count_gap = 0;
resample->count_nongap = 0;
resample->t0 = GST_CLOCK_TIME_NONE;
resample->in_offset0 = GST_BUFFER_OFFSET_NONE;
@ -957,9 +958,8 @@ gst_audio_resample_event (GstBaseTransform * base, GstEvent * event)
gst_audio_resample_push_drain (resample, resample->count_nongap);
gst_audio_resample_reset_state (resample);
if (resample->state)
resample->count_gap = resample->funcs->get_filt_len (resample->state);
else
resample->count_gap = 0;
resample->funcs->skip_zeros (resample->state);
resample->count_gap = 0;
resample->count_nongap = 0;
resample->t0 = GST_CLOCK_TIME_NONE;
resample->in_offset0 = GST_BUFFER_OFFSET_NONE;
@ -985,9 +985,6 @@ gst_audio_resample_check_discont (GstAudioResample * resample, GstBuffer * buf)
{
guint64 offset;
guint64 delta;
guint filt_len = resample->funcs->get_filt_len (resample->state);
guint64 delay =
gst_util_uint64_scale_round (filt_len, GST_SECOND, 2 * resample->inrate);
/* is the incoming buffer a discontinuity? */
if (G_UNLIKELY (GST_BUFFER_IS_DISCONT (buf)))
@ -1001,7 +998,7 @@ gst_audio_resample_check_discont (GstAudioResample * resample, GstBuffer * buf)
/* convert the inbound timestamp to an offset. */
offset =
gst_util_uint64_scale_int_round (GST_BUFFER_TIMESTAMP (buf) -
resample->t0 - delay, resample->inrate, GST_SECOND);
resample->t0, resample->inrate, GST_SECOND);
/* many elements generate imperfect streams due to rounding errors, so we
* permit a small error (up to one sample) without triggering a filter
@ -1054,9 +1051,12 @@ gst_audio_resample_process (GstAudioResample * resample, GstBuffer * inbuf,
{
guint num, den;
resample->funcs->get_ratio (resample->state, &num, &den);
out_processed =
gst_util_uint64_scale_int_ceil (resample->samples_in + in_len, den,
num) - resample->samples_out;
if (resample->samples_in + in_len >= filt_len / 2)
out_processed =
gst_util_uint64_scale_int_ceil (resample->samples_in + in_len -
filt_len / 2, den, num) - resample->samples_out;
else
out_processed = 0;
memset (GST_BUFFER_DATA (outbuf), 0, GST_BUFFER_SIZE (outbuf));
GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_GAP);
@ -1209,17 +1209,16 @@ gst_audio_resample_transform (GstBaseTransform * base, GstBuffer * inbuf,
/* handle discontinuity */
if (G_UNLIKELY (resample->need_discont)) {
guint filt_len = resample->funcs->get_filt_len (resample->state);
guint64 delay = gst_util_uint64_scale_round (filt_len, GST_SECOND,
2 * resample->inrate);
resample->count_gap = resample->funcs->get_filt_len (resample->state);
resample->funcs->skip_zeros (resample->state);
resample->count_gap = 0;
resample->count_nongap = 0;
/* reset */
resample->samples_in = 0;
resample->samples_out = 0;
GST_DEBUG_OBJECT (resample, "found discontinuity; resyncing");
/* resync the timestamp and offset counters if possible */
if (GST_BUFFER_TIMESTAMP_IS_VALID (inbuf)) {
resample->t0 = GST_BUFFER_TIMESTAMP (inbuf) - delay;
resample->t0 = GST_BUFFER_TIMESTAMP (inbuf);
} else {
GST_DEBUG_OBJECT (resample, "... but new timestamp is invalid");
resample->t0 = GST_CLOCK_TIME_NONE;