[gst-cvs] gst-plugins-base: audiortppay: fix frame duration calculations

[<wt...@ke...>]

Module: gst-plugins-base
Branch: master
Commit: 3a3c6f309c6b4da41a8e752310cdde8cfa80573c
URL:    http://cgit.freedesktop.org/gstreamer/gst-plugins-base/commit/?id=3a3c6f309c6b4da41a8e752310cdde8cfa80573c

Author: Wim Taymans <wim...@co...>
Date:   Thu Sep  3 14:13:44 2009 +0200

audiortppay: fix frame duration calculations

Fix the calculation of the frame duration and rtp timestamps.
Add some debugging

---

 gst-libs/gst/rtp/gstbasertpaudiopayload.c |   25 ++++++++++++++++++-------
 1 files changed, 18 insertions(+), 7 deletions(-)

diff --git a/gst-libs/gst/rtp/gstbasertpaudiopayload.c b/gst-libs/gst/rtp/gstbasertpaudiopayload.c
index f19552a..d6458a0 100644
--- a/gst-libs/gst/rtp/gstbasertpaudiopayload.c
+++ b/gst-libs/gst/rtp/gstbasertpaudiopayload.c
@@ -258,10 +258,13 @@ gst_base_rtp_audio_payload_set_frame_options (GstBaseRTPAudioPayload
 {
   g_return_if_fail (basertpaudiopayload != NULL);
 
-  basertpaudiopayload->frame_size = frame_size;
   basertpaudiopayload->frame_duration = frame_duration;
+  basertpaudiopayload->frame_size = frame_size;
 
   gst_adapter_clear (basertpaudiopayload->priv->adapter);
+
+  GST_DEBUG_OBJECT (basertpaudiopayload, "frame set to %d ms and size %d",
+      frame_duration, frame_size);
 }
 
 /**
@@ -308,6 +311,9 @@ gst_base_rtp_audio_payload_set_samplebits_options (GstBaseRTPAudioPayload
   basertpaudiopayload->priv->fragment_size = fragment_size / 8;
 
   gst_adapter_clear (basertpaudiopayload->priv->adapter);
+
+  GST_DEBUG_OBJECT (basertpaudiopayload,
+      "Samplebits set to sample size %d bits", sample_size);
 }
 
 static void
@@ -514,16 +520,21 @@ static GstClockTime
 gst_base_rtp_audio_payload_get_frame_duration (GstBaseRTPAudioPayload *
     payload, guint64 bytes)
 {
-  return gst_util_uint64_scale (bytes, payload->frame_duration * GST_MSECOND,
-      payload->frame_size);
+  return (bytes / payload->frame_size) * (payload->frame_duration *
+      GST_MSECOND);
 }
 
 static guint32
 gst_base_rtp_audio_payload_get_frame_rtptime (GstBaseRTPAudioPayload * payload,
     guint64 bytes)
 {
-  return gst_util_uint64_scale (bytes, payload->frame_duration * GST_MSECOND,
-      payload->frame_size * GST_BASE_RTP_PAYLOAD_CAST (payload)->clock_rate);
+  GstClockTime duration;
+
+  duration =
+      (bytes / payload->frame_size) * (payload->frame_duration * GST_MSECOND);
+
+  return gst_util_uint64_scale_int (duration,
+      GST_BASE_RTP_PAYLOAD_CAST (payload)->clock_rate, GST_SECOND);
 }
 
 static gboolean
@@ -575,8 +586,8 @@ static GstClockTime
 gst_base_rtp_audio_payload_get_sample_duration (GstBaseRTPAudioPayload *
     payload, guint64 bytes)
 {
-  return (bytes * 8 * GST_SECOND) /
-      (GST_BASE_RTP_PAYLOAD (payload)->clock_rate * payload->sample_size);
+  return gst_util_uint64_scale (bytes * 8, GST_SECOND,
+      GST_BASE_RTP_PAYLOAD (payload)->clock_rate * payload->sample_size);
 }
 
 static guint32