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baseaudioencoder: misc fixes

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Mark Nauwelaerts 2011-03-16 18:41:03 +01:00 committed by Tim-Philipp Müller
parent 8c61685554
commit 87409f2587
1 changed files with 12 additions and 9 deletions
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21
gst-libs/gst/audio/gstbaseaudioencoder.c
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@ -911,6 +911,13 @@ gst_base_audio_encoder_sink_setcaps (GstPad * pad, GstCaps * caps)
if (!gst_caps_is_fixed (caps)) if (!gst_caps_is_fixed (caps))
goto refuse_caps; goto refuse_caps;
/* adjust ts tracking to new sample rate */
if (GST_CLOCK_TIME_IS_VALID (enc->priv->base_ts) && state->rate) {
enc->priv->base_ts +=
GST_FRAMES_TO_CLOCK_TIME (enc->priv->samples, state->rate);
enc->priv->samples = 0;
}
if (!gst_base_audio_parse_caps (caps, state, &changed)) if (!gst_base_audio_parse_caps (caps, state, &changed))
goto refuse_caps; goto refuse_caps;
@ -1072,17 +1079,13 @@ gst_base_audio_encoder_sink_eventfunc (GstBaseAudioEncoder * enc,
if (format == GST_FORMAT_TIME) { if (format == GST_FORMAT_TIME) {
GST_DEBUG_OBJECT (enc, "received TIME NEW_SEGMENT %" GST_TIME_FORMAT GST_DEBUG_OBJECT (enc, "received TIME NEW_SEGMENT %" GST_TIME_FORMAT
" -- %" GST_TIME_FORMAT ", time %" GST_TIME_FORMAT " -- %" GST_TIME_FORMAT ", time %" GST_TIME_FORMAT
", accum %" GST_TIME_FORMAT, ", rate %g, applied_rate %g",
GST_TIME_ARGS (enc->segment.start), GST_TIME_ARGS (start), GST_TIME_ARGS (stop), GST_TIME_ARGS (time),
GST_TIME_ARGS (enc->segment.stop), rate, arate);
GST_TIME_ARGS (enc->segment.time),
GST_TIME_ARGS (enc->segment.accum));
} else { } else {
GST_DEBUG_OBJECT (enc, "received NEW_SEGMENT %" G_GINT64_FORMAT GST_DEBUG_OBJECT (enc, "received NEW_SEGMENT %" G_GINT64_FORMAT
" -- %" G_GINT64_FORMAT ", time %" G_GINT64_FORMAT " -- %" G_GINT64_FORMAT ", time %" G_GINT64_FORMAT
", accum %" G_GINT64_FORMAT, ", rate %g, applied_rate %g", start, stop, time, rate, arate);
enc->segment.start, enc->segment.stop,
enc->segment.time, enc->segment.accum);
GST_DEBUG_OBJECT (enc, "unsupported format; ignoring"); GST_DEBUG_OBJECT (enc, "unsupported format; ignoring");
break; break;
} }
@ -1102,7 +1105,7 @@ gst_base_audio_encoder_sink_eventfunc (GstBaseAudioEncoder * enc,
case GST_EVENT_FLUSH_STOP: case GST_EVENT_FLUSH_STOP:
/* discard any pending stuff */ /* discard any pending stuff */
// TODO route through drain ? /* TODO route through drain ?? */
if (!enc->priv->drained && klass->flush) if (!enc->priv->drained && klass->flush)
klass->flush (enc); klass->flush (enc);
/* and get (re)set for the sequel */ /* and get (re)set for the sequel */