static GstBuffer *
gst_rtp_g722_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf)
{
GstRtpG722Depay *rtpg722depay;
GstBuffer *outbuf;
gint payload_len;
gboolean marker;
rtpg722depay = GST_RTP_G722_DEPAY (depayload);
payload_len = gst_rtp_buffer_get_payload_len (buf);
if (payload_len <= 0)
goto empty_packet;
GST_DEBUG_OBJECT (rtpg722depay, "got payload of %d bytes", payload_len);
outbuf = gst_rtp_buffer_get_payload_buffer (buf);
marker = gst_rtp_buffer_get_marker (buf);
if (marker) {
/* mark talk spurt with DISCONT */
GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
}
return outbuf;
/* ERRORS */
empty_packet:
{
GST_ELEMENT_WARNING (rtpg722depay, STREAM, DECODE,
("Empty Payload."), (NULL));
return NULL;
}
}
static GstBuffer *
gst_rtp_g722_depay_process (GstRTPBaseDepayload * depayload, GstBuffer * buf)
{
GstRtpG722Depay *rtpg722depay;
GstBuffer *outbuf;
gint payload_len;
gboolean marker;
GstRTPBuffer rtp = { NULL };
rtpg722depay = GST_RTP_G722_DEPAY (depayload);
gst_rtp_buffer_map (buf, GST_MAP_READ, &rtp);
payload_len = gst_rtp_buffer_get_payload_len (&rtp);
if (payload_len <= 0)
goto empty_packet;
GST_DEBUG_OBJECT (rtpg722depay, "got payload of %d bytes", payload_len);
outbuf = gst_rtp_buffer_get_payload_buffer (&rtp);
marker = gst_rtp_buffer_get_marker (&rtp);
gst_rtp_buffer_unmap (&rtp);
if (marker && outbuf) {
/* mark talk spurt with RESYNC */
GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_RESYNC);
}
if (outbuf) {
gst_rtp_drop_meta (GST_ELEMENT_CAST (rtpg722depay), outbuf,
g_quark_from_static_string (GST_META_TAG_AUDIO_STR));
}
return outbuf;
/* ERRORS */
empty_packet:
{
GST_ELEMENT_WARNING (rtpg722depay, STREAM, DECODE,
("Empty Payload."), (NULL));
gst_rtp_buffer_unmap (&rtp);
return NULL;
}
}
static gboolean
gst_rtp_g722_depay_setcaps (GstBaseRTPDepayload * depayload, GstCaps * caps)
{
GstStructure *structure;
GstRtpG722Depay *rtpg722depay;
gint clock_rate, payload, samplerate;
gint channels;
GstCaps *srccaps;
gboolean res;
const gchar *channel_order;
const GstRTPChannelOrder *order;
rtpg722depay = GST_RTP_G722_DEPAY (depayload);
structure = gst_caps_get_structure (caps, 0);
payload = 96;
gst_structure_get_int (structure, "payload", &payload);
switch (payload) {
case GST_RTP_PAYLOAD_G722:
channels = 1;
clock_rate = 8000;
samplerate = 16000;
break;
default:
/* no fixed mapping, we need clock-rate */
channels = 0;
clock_rate = 0;
samplerate = 0;
break;
}
/* caps can overwrite defaults */
clock_rate =
gst_rtp_g722_depay_parse_int (structure, "clock-rate", clock_rate);
if (clock_rate == 0)
goto no_clockrate;
if (clock_rate == 8000)
samplerate = 16000;
if (samplerate == 0)
samplerate = clock_rate;
channels =
gst_rtp_g722_depay_parse_int (structure, "encoding-params", channels);
if (channels == 0) {
channels = gst_rtp_g722_depay_parse_int (structure, "channels", channels);
if (channels == 0) {
/* channels defaults to 1 otherwise */
channels = 1;
}
}
depayload->clock_rate = clock_rate;
rtpg722depay->rate = samplerate;
rtpg722depay->channels = channels;
srccaps = gst_caps_new_simple ("audio/G722",
"rate", G_TYPE_INT, samplerate, "channels", G_TYPE_INT, channels, NULL);
/* add channel positions */
channel_order = gst_structure_get_string (structure, "channel-order");
order = gst_rtp_channels_get_by_order (channels, channel_order);
if (order) {
gst_audio_set_channel_positions (gst_caps_get_structure (srccaps, 0),
order->pos);
} else {
GstAudioChannelPosition *pos;
GST_ELEMENT_WARNING (rtpg722depay, STREAM, DECODE,
(NULL), ("Unknown channel order '%s' for %d channels",
GST_STR_NULL (channel_order), channels));
/* create default NONE layout */
pos = gst_rtp_channels_create_default (channels);
gst_audio_set_channel_positions (gst_caps_get_structure (srccaps, 0), pos);
g_free (pos);
}
res = gst_pad_set_caps (depayload->srcpad, srccaps);
gst_caps_unref (srccaps);
return res;
/* ERRORS */
no_clockrate:
{
GST_ERROR_OBJECT (depayload, "no clock-rate specified");
return FALSE;
}
}