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audiortppayload: don't check adapter

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the adapter is never NULL so we don't need to check it.
Use _scale functions to avoid overflows.
This commit is contained in:
Wim Taymans 2009-09-01 18:08:14 +02:00
parent 4cacc441d8
commit 3c29efa692
1 changed files with 14 additions and 29 deletions
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43
gst-libs/gst/rtp/gstbasertpaudiopayload.c
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@ -233,9 +233,7 @@ gst_base_rtp_audio_payload_set_frame_options (GstBaseRTPAudioPayload
basertpaudiopayload->frame_size = frame_size;
basertpaudiopayload->frame_duration = frame_duration;
if (basertpaudiopayload->priv->adapter) {
gst_adapter_clear (basertpaudiopayload->priv->adapter);
}
gst_adapter_clear (basertpaudiopayload->priv->adapter);
}
/**
@ -255,9 +253,7 @@ gst_base_rtp_audio_payload_set_sample_options (GstBaseRTPAudioPayload
/* sample_size is in bits internally */
basertpaudiopayload->sample_size = sample_size * 8;
if (basertpaudiopayload->priv->adapter) {
gst_adapter_clear (basertpaudiopayload->priv->adapter);
}
gst_adapter_clear (basertpaudiopayload->priv->adapter);
}
/**
@ -277,9 +273,7 @@ gst_base_rtp_audio_payload_set_samplebits_options (GstBaseRTPAudioPayload
basertpaudiopayload->sample_size = sample_size;
if (basertpaudiopayload->priv->adapter) {
gst_adapter_clear (basertpaudiopayload->priv->adapter);
}
gst_adapter_clear (basertpaudiopayload->priv->adapter);
}
static GstFlowReturn
@ -374,8 +368,7 @@ gst_base_rtp_audio_payload_handle_frame_based_buffer (GstBaseRTPPayload *
"Calculated min_payload_len %u and max_payload_len %u",
min_payload_len, max_payload_len);
if (basertpaudiopayload->priv->adapter &&
gst_adapter_available (basertpaudiopayload->priv->adapter)) {
if (gst_adapter_available (basertpaudiopayload->priv->adapter)) {
/* If there is always data in the adapter, we have to use it */
gst_adapter_push (basertpaudiopayload->priv->adapter, buffer);
available = gst_adapter_available (basertpaudiopayload->priv->adapter);
@ -430,7 +423,7 @@ gst_base_rtp_audio_payload_handle_frame_based_buffer (GstBaseRTPPayload *
}
if (!use_adapter) {
if (available != 0 && basertpaudiopayload->priv->adapter) {
if (available != 0) {
GstBuffer *buf;
buf = gst_buffer_create_sub (buffer,
@ -452,7 +445,6 @@ gst_base_rtp_audio_payload_handle_sample_based_buffer (GstBaseRTPPayload *
const guint8 *data = NULL;
GstFlowReturn ret;
guint available;
guint maxptime_octets = G_MAXUINT;
guint minptime_octets = 0;
guint min_payload_len;
@ -479,8 +471,8 @@ gst_base_rtp_audio_payload_handle_sample_based_buffer (GstBaseRTPPayload *
/* max number of bytes based on given ptime */
if (basepayload->max_ptime != -1) {
maxptime_octets = 8 * basepayload->max_ptime * basepayload->clock_rate /
(basertpaudiopayload->sample_size * GST_SECOND);
maxptime_octets = gst_util_uint64_scale (basepayload->max_ptime * 8,
basepayload->clock_rate, basertpaudiopayload->sample_size * GST_SECOND);
}
max_payload_len = MIN (
@ -492,8 +484,8 @@ gst_base_rtp_audio_payload_handle_sample_based_buffer (GstBaseRTPPayload *
/* min number of bytes based on a given ptime, has to be a multiple
of sample rate */
minptime_octets = 8 * basepayload->min_ptime * basepayload->clock_rate /
(basertpaudiopayload->sample_size * GST_SECOND);
minptime_octets = gst_util_uint64_scale (basepayload->min_ptime * 8,
basepayload->clock_rate, basertpaudiopayload->sample_size * GST_SECOND);
min_payload_len = MAX (minptime_octets, fragment_size);
@ -505,8 +497,7 @@ gst_base_rtp_audio_payload_handle_sample_based_buffer (GstBaseRTPPayload *
"Calculated min_payload_len %u and max_payload_len %u",
min_payload_len, max_payload_len);
if (basertpaudiopayload->priv->adapter &&
gst_adapter_available (basertpaudiopayload->priv->adapter)) {
if (gst_adapter_available (basertpaudiopayload->priv->adapter)) {
/* If there is always data in the adapter, we have to use it */
gst_adapter_push (basertpaudiopayload->priv->adapter, buffer);
available = gst_adapter_available (basertpaudiopayload->priv->adapter);
@ -564,7 +555,7 @@ gst_base_rtp_audio_payload_handle_sample_based_buffer (GstBaseRTPPayload *
}
if (!use_adapter) {
if (available != 0 && basertpaudiopayload->priv->adapter) {
if (available != 0) {
GstBuffer *buf;
buf = gst_buffer_create_sub (buffer,
@ -633,9 +624,7 @@ gst_base_rtp_payload_audio_change_state (GstElement * element,
switch (transition) {
case GST_STATE_CHANGE_PAUSED_TO_READY:
if (basertppayload->priv->adapter) {
gst_adapter_clear (basertppayload->priv->adapter);
}
gst_adapter_clear (basertppayload->priv->adapter);
break;
default:
break;
@ -654,14 +643,10 @@ gst_base_rtp_payload_audio_handle_event (GstPad * pad, GstEvent * event)
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_EOS:
if (basertpaudiopayload->priv->adapter) {
gst_adapter_clear (basertpaudiopayload->priv->adapter);
}
gst_adapter_clear (basertpaudiopayload->priv->adapter);
break;
case GST_EVENT_FLUSH_STOP:
if (basertpaudiopayload->priv->adapter) {
gst_adapter_clear (basertpaudiopayload->priv->adapter);
}
gst_adapter_clear (basertpaudiopayload->priv->adapter);
break;
default:
break;