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rtpg729pay: Simplify adapter usage

Browse source 
https://bugzilla.gnome.org/show_bug.cgi?id=606050
This commit is contained in:
Olivier Crête 2009-12-31 16:52:30 -05:00
parent 0a18587792
commit 63a9db5826
1 changed files with 20 additions and 47 deletions
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67
gst/rtp/gstrtpg729pay.c
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@ -156,7 +156,6 @@ gst_rtp_g729_pay_handle_buffer (GstBaseRTPPayload * payload, GstBuffer * buf)
guint minptime_octets = 0;
guint min_payload_len;
guint max_payload_len;
gboolean use_adapter = FALSE;
available = GST_BUFFER_SIZE (buf);
@ -221,31 +220,25 @@ gst_rtp_g729_pay_handle_buffer (GstBaseRTPPayload * payload, GstBuffer * buf)
adapter = gst_base_rtp_audio_payload_get_adapter (basertpaudiopayload);
if (adapter && gst_adapter_available (adapter)) {
/* If there is always data in the adapter, we have to use it */
gst_adapter_push (adapter, buf);
available = gst_adapter_available (adapter);
use_adapter = TRUE;
} else {
/* let's set the base timestamp */
/* let's reset the base timestamp when the adapter is empty */
if (gst_adapter_available (adapter) == 0)
basertpaudiopayload->base_ts = GST_BUFFER_TIMESTAMP (buf);
/* If buffer fits on an RTP packet, let's just push it through */
/* this will check against max_ptime and max_mtu */
if (GST_BUFFER_SIZE (buf) >= min_payload_len &&
GST_BUFFER_SIZE (buf) <= max_payload_len) {
ret = gst_base_rtp_audio_payload_push (basertpaudiopayload,
GST_BUFFER_DATA (buf), GST_BUFFER_SIZE (buf),
GST_BUFFER_TIMESTAMP (buf));
gst_buffer_unref (buf);
g_object_unref (adapter);
return ret;
}
available = GST_BUFFER_SIZE (buf);
data = (guint8 *) GST_BUFFER_DATA (buf);
if (gst_adapter_available (adapter) == 0 &&
GST_BUFFER_SIZE (buf) >= min_payload_len &&
GST_BUFFER_SIZE (buf) <= max_payload_len) {
ret = gst_base_rtp_audio_payload_push (basertpaudiopayload,
GST_BUFFER_DATA (buf), GST_BUFFER_SIZE (buf),
GST_BUFFER_TIMESTAMP (buf));
gst_buffer_unref (buf);
g_object_unref (adapter);
return ret;
}
gst_adapter_push (adapter, buf);
available = gst_adapter_available (adapter);
/* as long as we have full frames */
/* this loop will push all available buffers till the last frame */
while (available >= min_payload_len ||
@ -260,38 +253,18 @@ gst_rtp_g729_pay_handle_buffer (GstBaseRTPPayload * payload, GstBuffer * buf)
(available / G729_FRAME_SIZE) * G729_FRAME_SIZE);
}
if (use_adapter) {
data = gst_adapter_peek (adapter, payload_len);
}
data = gst_adapter_peek (adapter, payload_len);
ret = gst_base_rtp_audio_payload_push (basertpaudiopayload, data,
payload_len, basertpaudiopayload->base_ts);
ret = gst_base_rtp_audio_payload_flush (basertpaudiopayload, payload_len,
basertpaudiopayload->base_ts);
num = payload_len / G729_FRAME_SIZE;
basertpaudiopayload->base_ts += G729_FRAME_DURATION * num;
if (use_adapter) {
gst_adapter_flush (adapter, payload_len);
available = gst_adapter_available (adapter);
} else {
available -= payload_len;
data += payload_len;
}
available = gst_adapter_available (adapter);
}
if (!use_adapter) {
if (available != 0 && adapter) {
GstBuffer *buf2;
buf2 = gst_buffer_create_sub (buf,
GST_BUFFER_SIZE (buf) - available, available);
gst_adapter_push (adapter, buf2);
}
gst_buffer_unref (buf);
}
if (adapter) {
g_object_unref (adapter);
}
g_object_unref (adapter);
return ret;