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rtpbaseaudiopayload: Use GST_BUFFER_PTS() instead of GST_BUFFER_TIMESTAMP()

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Sebastian Dröge 2015-06-10 14:33:01 +02:00
parent 7b78a33dc6
commit abcaa71485
1 changed files with 2 additions and 2 deletions
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4
gst-libs/gst/rtp/gstrtpbaseaudiopayload.c
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@ -415,7 +415,7 @@ gst_rtp_base_audio_payload_set_meta (GstRTPBaseAudioPayload * payload,
} }
gst_rtp_buffer_unmap (&rtp); gst_rtp_buffer_unmap (&rtp);
GST_BUFFER_TIMESTAMP (buffer) = timestamp; GST_BUFFER_PTS (buffer) = timestamp;
/* get the offset in RTP time */ /* get the offset in RTP time */
GST_BUFFER_OFFSET (buffer) = priv->bytes_to_rtptime (payload, priv->offset); GST_BUFFER_OFFSET (buffer) = priv->bytes_to_rtptime (payload, priv->offset);
@ -820,7 +820,7 @@ gst_rtp_base_audio_payload_handle_buffer (GstRTPBasePayload *
payload = GST_RTP_BASE_AUDIO_PAYLOAD_CAST (basepayload); payload = GST_RTP_BASE_AUDIO_PAYLOAD_CAST (basepayload);
priv = payload->priv; priv = payload->priv;
timestamp = GST_BUFFER_TIMESTAMP (buffer); timestamp = GST_BUFFER_PTS (buffer);
discont = GST_BUFFER_IS_DISCONT (buffer); discont = GST_BUFFER_IS_DISCONT (buffer);
if (discont) { if (discont) {