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audiortppay: fix frame duration calculations

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Fix the calculation of the frame duration and rtp timestamps.
Add some debugging
This commit is contained in:
Wim Taymans 2009-09-03 14:13:44 +02:00
parent bfc19462bb
commit 3a3c6f309c
1 changed files with 18 additions and 7 deletions
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25
gst-libs/gst/rtp/gstbasertpaudiopayload.c
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@ -258,10 +258,13 @@ gst_base_rtp_audio_payload_set_frame_options (GstBaseRTPAudioPayload
{ {
g_return_if_fail (basertpaudiopayload != NULL); g_return_if_fail (basertpaudiopayload != NULL);
basertpaudiopayload->frame_size = frame_size;
basertpaudiopayload->frame_duration = frame_duration; basertpaudiopayload->frame_duration = frame_duration;
basertpaudiopayload->frame_size = frame_size;
gst_adapter_clear (basertpaudiopayload->priv->adapter); 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; basertpaudiopayload->priv->fragment_size = fragment_size / 8;
gst_adapter_clear (basertpaudiopayload->priv->adapter); gst_adapter_clear (basertpaudiopayload->priv->adapter);
GST_DEBUG_OBJECT (basertpaudiopayload,
"Samplebits set to sample size %d bits", sample_size);
} }
static void static void
@ -514,16 +520,21 @@ static GstClockTime
gst_base_rtp_audio_payload_get_frame_duration (GstBaseRTPAudioPayload * gst_base_rtp_audio_payload_get_frame_duration (GstBaseRTPAudioPayload *
payload, guint64 bytes) payload, guint64 bytes)
{ {
return gst_util_uint64_scale (bytes, payload->frame_duration * GST_MSECOND, return (bytes / payload->frame_size) * (payload->frame_duration *
payload->frame_size); GST_MSECOND);
} }
static guint32 static guint32
gst_base_rtp_audio_payload_get_frame_rtptime (GstBaseRTPAudioPayload * payload, gst_base_rtp_audio_payload_get_frame_rtptime (GstBaseRTPAudioPayload * payload,
guint64 bytes) guint64 bytes)
{ {
return gst_util_uint64_scale (bytes, payload->frame_duration * GST_MSECOND, GstClockTime duration;
payload->frame_size * GST_BASE_RTP_PAYLOAD_CAST (payload)->clock_rate);
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 static gboolean
@ -575,8 +586,8 @@ static GstClockTime
gst_base_rtp_audio_payload_get_sample_duration (GstBaseRTPAudioPayload * gst_base_rtp_audio_payload_get_sample_duration (GstBaseRTPAudioPayload *
payload, guint64 bytes) payload, guint64 bytes)
{ {
return (bytes * 8 * GST_SECOND) / return gst_util_uint64_scale (bytes * 8, GST_SECOND,
(GST_BASE_RTP_PAYLOAD (payload)->clock_rate * payload->sample_size); GST_BASE_RTP_PAYLOAD (payload)->clock_rate * payload->sample_size);
} }
static guint32 static guint32