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audiortppay: add some debugging

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This commit is contained in:
Wim Taymans 2009-09-03 18:53:19 +02:00
parent c1db9ebb20
commit 7a7663476f
1 changed files with 10 additions and 2 deletions
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12
gst-libs/gst/rtp/gstbasertpaudiopayload.c
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@ -668,11 +668,19 @@ gst_base_rtp_audio_payload_handle_buffer (GstBaseRTPPayload *
* output buffers containing the same RTP timestamp gap as the gap * output buffers containing the same RTP timestamp gap as the gap
* between the GST timestamps. */ * between the GST timestamps. */
if (timestamp > priv->last_timestamp) { if (timestamp > priv->last_timestamp) {
GstClockTime diff;
guint64 bytes;
/* we're only going to apply a positive gap, otherwise we let the marker /* we're only going to apply a positive gap, otherwise we let the marker
* bit do its thing. simply convert to bytes and add the the current * bit do its thing. simply convert to bytes and add the the current
* offset */ * offset */
priv->offset += diff = timestamp - priv->last_timestamp;
priv->time_to_bytes (payload, timestamp - priv->last_timestamp); bytes = priv->time_to_bytes (payload, diff);
priv->offset += bytes;
GST_DEBUG_OBJECT (payload,
"elapsed time %" GST_TIME_FORMAT ", bytes %" G_GUINT64_FORMAT
", new offset %" G_GUINT64_FORMAT, GST_TIME_ARGS (diff), bytes,
priv->offset);
} }
} }
} }