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audiopayload: code cleanups

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This commit is contained in:
Wim Taymans 2009-09-01 18:26:52 +02:00
parent 3c29efa692
commit bc3c8a1564
1 changed files with 32 additions and 22 deletions
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52
gst-libs/gst/rtp/gstbasertpaudiopayload.c
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@ -311,7 +311,7 @@ gst_base_rtp_audio_payload_handle_frame_based_buffer (GstBaseRTPPayload *
GstFlowReturn ret;
guint available;
gint frame_size, frame_duration;
guint max_frames;
guint maxptime_octets = G_MAXUINT;
guint minptime_octets = 0;
guint min_payload_len;
@ -324,11 +324,9 @@ gst_base_rtp_audio_payload_handle_frame_based_buffer (GstBaseRTPPayload *
basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (basepayload);
if (basertpaudiopayload->frame_size == 0 ||
basertpaudiopayload->frame_duration == 0) {
GST_DEBUG_OBJECT (basertpaudiopayload, "Required options not set");
gst_buffer_unref (buffer);
return GST_FLOW_ERROR;
}
basertpaudiopayload->frame_duration == 0)
goto config_error;
frame_size = basertpaudiopayload->frame_size;
frame_duration = basertpaudiopayload->frame_duration;
@ -345,12 +343,13 @@ gst_base_rtp_audio_payload_handle_frame_based_buffer (GstBaseRTPPayload *
}
}
max_payload_len = MIN (
/* MTU max */
(int) (gst_rtp_buffer_calc_payload_len (GST_BASE_RTP_PAYLOAD_MTU
(basertpaudiopayload), 0, 0) / frame_size) * frame_size,
/* ptime max */
maxptime_octets);
max_frames = gst_rtp_buffer_calc_payload_len (GST_BASE_RTP_PAYLOAD_MTU
(basertpaudiopayload), 0, 0);
/* round down to frame_size */
max_frames = (max_frames / frame_size) * frame_size;
/* max payload length */
max_payload_len = MIN (max_frames, maxptime_octets);
/* min number of bytes based on a given ptime, has to be a multiple
of frame duration */
@ -360,9 +359,8 @@ gst_base_rtp_audio_payload_handle_frame_based_buffer (GstBaseRTPPayload *
min_payload_len = MAX (minptime_octets, frame_size);
if (min_payload_len > max_payload_len) {
if (min_payload_len > max_payload_len)
min_payload_len = max_payload_len;
}
GST_DEBUG_OBJECT (basertpaudiopayload,
"Calculated min_payload_len %u and max_payload_len %u",
@ -432,8 +430,15 @@ gst_base_rtp_audio_payload_handle_frame_based_buffer (GstBaseRTPPayload *
}
gst_buffer_unref (buffer);
}
return ret;
/* ERRORS */
config_error:
{
GST_DEBUG_OBJECT (basertpaudiopayload, "Required options not set");
gst_buffer_unref (buffer);
return GST_FLOW_ERROR;
}
}
static GstFlowReturn
@ -457,11 +462,8 @@ gst_base_rtp_audio_payload_handle_sample_based_buffer (GstBaseRTPPayload *
basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (basepayload);
if (basertpaudiopayload->sample_size == 0) {
GST_DEBUG_OBJECT (basertpaudiopayload, "Required options not set");
gst_buffer_unref (buffer);
return GST_FLOW_ERROR;
}
if (basertpaudiopayload->sample_size == 0)
goto config_error;
/* sample_size is in bits and is converted into multiple bytes */
fragment_size = basertpaudiopayload->sample_size;
@ -489,9 +491,8 @@ gst_base_rtp_audio_payload_handle_sample_based_buffer (GstBaseRTPPayload *
min_payload_len = MAX (minptime_octets, fragment_size);
if (min_payload_len > max_payload_len) {
if (min_payload_len > max_payload_len)
min_payload_len = max_payload_len;
}
GST_DEBUG_OBJECT (basertpaudiopayload,
"Calculated min_payload_len %u and max_payload_len %u",
@ -566,6 +567,14 @@ gst_base_rtp_audio_payload_handle_sample_based_buffer (GstBaseRTPPayload *
}
return ret;
/* ERRORS */
config_error:
{
GST_DEBUG_OBJECT (basertpaudiopayload, "Required options not set");
gst_buffer_unref (buffer);
return GST_FLOW_ERROR;
}
}
/**
@ -643,6 +652,7 @@ gst_base_rtp_payload_audio_handle_event (GstPad * pad, GstEvent * event)
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_EOS:
/* FIXME. push remaining bytes */
gst_adapter_clear (basertpaudiopayload->priv->adapter);
break;
case GST_EVENT_FLUSH_STOP: