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audioresample: Drain the resampler and reset timestamp tracking on caps changes

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Especially when changing the sample rate our timestamp tracking will be
completely off, but even otherwise we would usually lose the last few
samples if we don't drain here as the resampler gets reset if anything
but the sample rate changes.

This is usually not a problem as the first buffer after a caps event
usually has the discont flag set, but can cause problems if
 - the caps event is followed by a segment event, which then causes
   draining according to the new sample rate
 - the caps were changed because of rengotiation due to a reconfigure
   event and there is not discontinuity from upstream

In both cases we would output buffers with completely wrong timestamps.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/670>
This commit is contained in:
Sebastian Dröge 2020-05-26 22:45:41 +03:00 committed by GStreamer Merge Bot
parent c7a956aede
commit 6d423cbba2
1 changed files with 24 additions and 1 deletions
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25
gst/audioresample/gstaudioresample.c
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@ -127,6 +127,9 @@ static gboolean gst_audio_resample_stop (GstBaseTransform * base);
static gboolean gst_audio_resample_query (GstPad * pad, GstObject * parent,
GstQuery * query);
static void gst_audio_resample_push_drain (GstAudioResample * resample,
guint history_len);
#define gst_audio_resample_parent_class parent_class
G_DEFINE_TYPE (GstAudioResample, gst_audio_resample, GST_TYPE_BASE_TRANSFORM);
@ -497,7 +500,27 @@ gst_audio_resample_set_caps (GstBaseTransform * base, GstCaps * incaps,
if (!gst_audio_info_from_caps (&out, outcaps))
goto invalid_outcaps;
/* FIXME do some checks */
/* Reset timestamp tracking and drain the resampler if the audio format is
* changing. Especially when changing the sample rate our timestamp tracking
* will be completely off, but even otherwise we would usually lose the last
* few samples if we don't drain here */
if (!gst_audio_info_is_equal (&in, &resample->in) ||
!gst_audio_info_is_equal (&out, &resample->out)) {
if (resample->converter) {
gsize latency = gst_audio_converter_get_max_latency (resample->converter);
gst_audio_resample_push_drain (resample, latency);
}
gst_audio_resample_reset_state (resample);
resample->num_gap_samples = 0;
resample->num_nongap_samples = 0;
resample->t0 = GST_CLOCK_TIME_NONE;
resample->in_offset0 = GST_BUFFER_OFFSET_NONE;
resample->out_offset0 = GST_BUFFER_OFFSET_NONE;
resample->samples_in = 0;
resample->samples_out = 0;
resample->need_discont = TRUE;
}
gst_audio_resample_update_state (resample, &in, &out);
resample->in = in;