else
order = NULL;
- gst_basertppayload_set_options (basepayload, "audio", TRUE, "L16", rate);
+ gst_base_rtp_payload_set_options (basepayload, "audio", TRUE, "L16", rate);
params = g_strdup_printf ("%d", channels);
if (!order && channels > 2) {
}
if (order && order->name) {
- res = gst_basertppayload_set_outcaps (basepayload,
+ res = gst_base_rtp_payload_set_outcaps (basepayload,
"encoding-params", G_TYPE_STRING, params, "channels", G_TYPE_INT,
channels, "channel-order", G_TYPE_STRING, order->name, NULL);
} else {
- res = gst_basertppayload_set_outcaps (basepayload,
+ res = gst_base_rtp_payload_set_outcaps (basepayload,
"encoding-params", G_TYPE_STRING, params, "channels", G_TYPE_INT,
channels, NULL);
}
if (!gst_structure_get_int (structure, "rate", &rate))
rate = 90000; /* default */
- gst_basertppayload_set_options (payload, "audio", TRUE, "AC3", rate);
- res = gst_basertppayload_set_outcaps (payload, NULL);
+ gst_base_rtp_payload_set_options (payload, "audio", TRUE, "AC3", rate);
+ res = gst_base_rtp_payload_set_outcaps (payload, NULL);
return res;
}
GST_BUFFER_TIMESTAMP (outbuf) = rtpac3pay->first_ts;
GST_BUFFER_DURATION (outbuf) = rtpac3pay->duration;
- ret = gst_basertppayload_push (GST_BASE_RTP_PAYLOAD (rtpac3pay), outbuf);
+ ret = gst_base_rtp_payload_push (GST_BASE_RTP_PAYLOAD (rtpac3pay), outbuf);
}
return ret;
/* if this buffer is going to overflow the packet, flush what we
* have. */
- if (gst_basertppayload_is_filled (basepayload,
+ if (gst_base_rtp_payload_is_filled (basepayload,
packet_len, rtpac3pay->duration + duration)) {
ret = gst_rtp_ac3_pay_flush (rtpac3pay);
avail = 0;
goto wrong_type;
if (rtpamrpay->mode == GST_RTP_AMR_P_MODE_NB)
- gst_basertppayload_set_options (basepayload, "audio", TRUE, "AMR", 8000);
+ gst_base_rtp_payload_set_options (basepayload, "audio", TRUE, "AMR", 8000);
else
- gst_basertppayload_set_options (basepayload, "audio", TRUE, "AMR-WB",
+ gst_base_rtp_payload_set_options (basepayload, "audio", TRUE, "AMR-WB",
16000);
- res = gst_basertppayload_set_outcaps (basepayload,
+ res = gst_base_rtp_payload_set_outcaps (basepayload,
"encoding-params", G_TYPE_STRING, "1", "octet-align", G_TYPE_STRING, "1",
/* don't set the defaults
*
gst_rtp_buffer_unmap (&rtp);
- ret = gst_basertppayload_push (basepayload, outbuf);
+ ret = gst_base_rtp_payload_push (basepayload, outbuf);
return ret;
goto wrong_mode;
if (mode == 16) {
- gst_basertppayload_set_options (basertppayload, "audio", TRUE, "BV16",
+ gst_base_rtp_payload_set_options (basertppayload, "audio", TRUE, "BV16",
8000);
basertppayload->clock_rate = 8000;
} else {
- gst_basertppayload_set_options (basertppayload, "audio", TRUE, "BV32",
+ gst_base_rtp_payload_set_options (basertppayload, "audio", TRUE, "BV32",
16000);
basertppayload->clock_rate = 16000;
}
payload = GST_BASE_RTP_PAYLOAD (rtpceltpay);
- gst_basertppayload_set_options (payload, "audio", FALSE, "CELT", rate);
+ gst_base_rtp_payload_set_options (payload, "audio", FALSE, "CELT", rate);
cstr = g_strdup_printf ("%d", nb_channels);
fsstr = g_strdup_printf ("%d", frame_size);
- res = gst_basertppayload_set_outcaps (payload, "encoding-params",
+ res = gst_base_rtp_payload_set_outcaps (payload, "encoding-params",
G_TYPE_STRING, cstr, "frame-size", G_TYPE_STRING, fsstr, NULL);
g_free (cstr);
g_free (fsstr);
rtpceltpay->sbytes = 0;
rtpceltpay->qduration = 0;
- ret = gst_basertppayload_push (GST_BASE_RTP_PAYLOAD (rtpceltpay), outbuf);
+ ret = gst_base_rtp_payload_push (GST_BASE_RTP_PAYLOAD (rtpceltpay), outbuf);
return ret;
}
packet_len = gst_rtp_buffer_calc_packet_len (payload_len, 0, 0);
- if (gst_basertppayload_is_filled (basepayload, packet_len, packet_dur)) {
+ if (gst_base_rtp_payload_is_filled (basepayload, packet_len, packet_dur)) {
/* size or duration would overflow the packet, flush the queued data */
ret = gst_rtp_celt_pay_flush_queued (rtpceltpay);
}
default:
break;
}
- gst_basertppayload_set_options (GST_BASE_RTP_PAYLOAD (rtpdvpay), media, TRUE,
- "DV", 90000);
+ gst_base_rtp_payload_set_options (GST_BASE_RTP_PAYLOAD (rtpdvpay), media,
+ TRUE, "DV", 90000);
if (audio_bundled) {
- res = gst_basertppayload_set_outcaps (GST_BASE_RTP_PAYLOAD (rtpdvpay),
+ res = gst_base_rtp_payload_set_outcaps (GST_BASE_RTP_PAYLOAD (rtpdvpay),
"encode", G_TYPE_STRING, encode,
"audio", G_TYPE_STRING, "bundled", NULL);
} else {
- res = gst_basertppayload_set_outcaps (GST_BASE_RTP_PAYLOAD (rtpdvpay),
+ res = gst_base_rtp_payload_set_outcaps (GST_BASE_RTP_PAYLOAD (rtpdvpay),
"encode", G_TYPE_STRING, encode, NULL);
}
return res;
/* Push out the created piece, and check for errors. */
gst_rtp_buffer_unmap (&rtp);
- ret = gst_basertppayload_push (basepayload, outbuf);
+ ret = gst_base_rtp_payload_push (basepayload, outbuf);
if (ret != GST_FLOW_OK)
break;
* RFC 3551 although the sampling rate is 16000 Hz */
clock_rate = 8000;
- gst_basertppayload_set_options (basepayload, "audio", TRUE, "G722",
+ gst_base_rtp_payload_set_options (basepayload, "audio", TRUE, "G722",
clock_rate);
params = g_strdup_printf ("%d", channels);
}
if (order && order->name) {
- res = gst_basertppayload_set_outcaps (basepayload,
+ res = gst_base_rtp_payload_set_outcaps (basepayload,
"encoding-params", G_TYPE_STRING, params, "channels", G_TYPE_INT,
channels, "channel-order", G_TYPE_STRING, order->name, NULL);
} else {
- res = gst_basertppayload_set_outcaps (basepayload,
+ res = gst_base_rtp_payload_set_outcaps (basepayload,
"encoding-params", G_TYPE_STRING, params, "channels", G_TYPE_INT,
channels, NULL);
}
pay->adapter = gst_adapter_new ();
payload->pt = GST_RTP_PAYLOAD_G723;
- gst_basertppayload_set_options (payload, "audio", FALSE, "G723", 8000);
+ gst_base_rtp_payload_set_options (payload, "audio", FALSE, "G723", 8000);
}
static void
payload->pt = pt;
payload->dynamic = pt != GST_RTP_PAYLOAD_G723;
- res = gst_basertppayload_set_outcaps (payload, NULL);
+ res = gst_base_rtp_payload_set_outcaps (payload, NULL);
return res;
}
}
gst_rtp_buffer_unmap (&rtp);
- ret = gst_basertppayload_push (GST_BASE_RTP_PAYLOAD (pay), outbuf);
+ ret = gst_base_rtp_payload_push (GST_BASE_RTP_PAYLOAD (pay), outbuf);
return ret;
}
packet_dur = pay->duration + G723_FRAME_DURATION;
packet_len = gst_rtp_buffer_calc_packet_len (payload_len, 0, 0);
- if (gst_basertppayload_is_filled (payload, packet_len, packet_dur)) {
+ if (gst_base_rtp_payload_is_filled (payload, packet_len, packet_dur)) {
/* size or duration would overflow the packet, flush the queued data */
ret = gst_rtp_g723_pay_flush (pay);
}
GST_DEBUG_OBJECT (payload, "no peer caps, AAL2 %d", pay->aal2);
}
- gst_basertppayload_set_options (payload, "audio", TRUE, encoding_name, 8000);
- res = gst_basertppayload_set_outcaps (payload, NULL);
+ gst_base_rtp_payload_set_options (payload, "audio", TRUE, encoding_name,
+ 8000);
+ res = gst_base_rtp_payload_set_outcaps (payload, NULL);
g_free (encoding_name);
GstBaseRTPPayload *payload = GST_BASE_RTP_PAYLOAD (pay);
payload->pt = GST_RTP_PAYLOAD_G729;
- gst_basertppayload_set_options (payload, "audio", FALSE, "G729", 8000);
+ gst_base_rtp_payload_set_options (payload, "audio", FALSE, "G729", 8000);
pay->adapter = gst_adapter_new ();
}
payload->pt = pt;
payload->dynamic = pt != GST_RTP_PAYLOAD_G729;
- res = gst_basertppayload_set_outcaps (payload, NULL);
+ res = gst_base_rtp_payload_set_outcaps (payload, NULL);
return res;
}
}
gst_rtp_buffer_unmap (&rtp);
- ret = gst_basertppayload_push (basepayload, outbuf);
+ ret = gst_base_rtp_payload_push (basepayload, outbuf);
return ret;
}
if (strcmp ("audio/x-gsm", stname))
goto invalid_type;
- gst_basertppayload_set_options (payload, "audio", FALSE, "GSM", 8000);
- res = gst_basertppayload_set_outcaps (payload, NULL);
+ gst_base_rtp_payload_set_options (payload, "audio", FALSE, "GSM", 8000);
+ res = gst_base_rtp_payload_set_outcaps (payload, NULL);
return res;
GST_DEBUG ("gst_rtp_gsm_pay_chain: pushing buffer of size %d",
gst_buffer_get_size (outbuf));
- ret = gst_basertppayload_push (basepayload, outbuf);
+ ret = gst_base_rtp_payload_push (basepayload, outbuf);
return ret;
capsenc = g_base64_encode ((guchar *) capsstr, strlen (capsstr));
g_free (capsstr);
- gst_basertppayload_set_options (payload, "application", TRUE, "X-GST", 90000);
+ gst_base_rtp_payload_set_options (payload, "application", TRUE, "X-GST",
+ 90000);
res =
- gst_basertppayload_set_outcaps (payload, "caps", G_TYPE_STRING, capsenc,
+ gst_base_rtp_payload_set_outcaps (payload, "caps", G_TYPE_STRING, capsenc,
NULL);
g_free (capsenc);
GST_BUFFER_TIMESTAMP (outbuf) = timestamp;
- ret = gst_basertppayload_push (basepayload, outbuf);
+ ret = gst_base_rtp_payload_push (basepayload, outbuf);
}
gst_buffer_unmap (buffer, data, size);
gst_buffer_unref (buffer);
if (!F && payload_len > 4 && (GST_READ_UINT32_BE (payload) >> 10 == 0x20)) {
GST_DEBUG ("Mode A with PSC => frame start");
rtph263depay->start = TRUE;
- if (!!(payload[4] & 0x02) != I) {
+ if (! !(payload[4] & 0x02) != I) {
GST_DEBUG ("Wrong Picture Coding Type Flag in rtp header");
I = !I;
}
if (rtph263depay->start) {
/* frame is completed */
guint avail;
- guint32 timestamp;
if (rtph263depay->offset) {
/* push in the leftover */
GST_DEBUG ("Pushing out a buffer of %d bytes", avail);
- timestamp = gst_rtp_buffer_get_timestamp (&rtp);
- gst_base_rtp_depayload_push_ts (depayload, timestamp, outbuf);
+ gst_base_rtp_depayload_push (depayload, outbuf);
rtph263depay->offset = 0;
rtph263depay->leftover = 0;
rtph263depay->start = FALSE;
gboolean res;
payload->pt = GST_RTP_PAYLOAD_H263;
- gst_basertppayload_set_options (payload, "video", TRUE, "H263", 90000);
- res = gst_basertppayload_set_outcaps (payload, NULL);
+ gst_base_rtp_payload_set_options (payload, "video", TRUE, "H263", 90000);
+ res = gst_base_rtp_payload_set_outcaps (payload, NULL);
return res;
}
gst_rtp_buffer_unmap (&rtp);
ret =
- gst_basertppayload_push (GST_BASE_RTP_PAYLOAD (rtph263pay),
+ gst_base_rtp_payload_push (GST_BASE_RTP_PAYLOAD (rtph263pay),
package->outbuf);
GST_DEBUG ("Package pushed, returning");
if (!encoding_name)
encoding_name = g_strdup ("H263-1998");
- gst_basertppayload_set_options (payload, "video", TRUE,
+ gst_base_rtp_payload_set_options (payload, "video", TRUE,
(gchar *) encoding_name, 90000);
- res = gst_basertppayload_set_outcaps (payload, NULL);
+ res = gst_base_rtp_payload_set_outcaps (payload, NULL);
g_free (encoding_name);
return res;
gst_adapter_flush (rtph263ppay->adapter, towrite);
- ret = gst_basertppayload_push (GST_BASE_RTP_PAYLOAD (rtph263ppay), outbuf);
+ ret =
+ gst_base_rtp_payload_push (GST_BASE_RTP_PAYLOAD (rtph263ppay), outbuf);
avail -= towrite;
fragmented = TRUE;
GstBuffer * buffer);
static gboolean gst_rtp_h264_pay_handle_event (GstBaseRTPPayload * payload,
GstEvent * event);
-static GstStateChangeReturn gst_basertppayload_change_state (GstElement *
+static GstStateChangeReturn gst_rtp_h264_pay_change_state (GstElement *
element, GstStateChange transition);
#define gst_rtp_h264_pay_parent_class parent_class
"Laurent Glayal <spglegle@yahoo.fr>");
gstelement_class->change_state =
- GST_DEBUG_FUNCPTR (gst_basertppayload_change_state);
+ GST_DEBUG_FUNCPTR (gst_rtp_h264_pay_change_state);
gstbasertppayload_class->get_caps = gst_rtp_h264_pay_getcaps;
gstbasertppayload_class->set_caps = gst_rtp_h264_pay_setcaps;
/* profile is 24 bit. Force it to respect the limit */
profile = g_strdup_printf ("%06x", payloader->profile & 0xffffff);
/* combine into output caps */
- res = gst_basertppayload_set_outcaps (basepayload,
+ res = gst_base_rtp_payload_set_outcaps (basepayload,
"sprop-parameter-sets", G_TYPE_STRING, sprops->str, NULL);
g_string_free (sprops, TRUE);
g_free (profile);
/* we can only set the output caps when we found the sprops and profile
* NALs */
- gst_basertppayload_set_options (basepayload, "video", TRUE, "H264", 90000);
+ gst_base_rtp_payload_set_options (basepayload, "video", TRUE, "H264", 90000);
alignment = gst_structure_get_string (str, "alignment");
if (alignment && !strcmp (alignment, "au"))
gst_buffer_list_add (list, paybuf);
/* push the list to the next element in the pipe */
- ret = gst_basertppayload_push_list (basepayload, list);
+ ret = gst_base_rtp_payload_push_list (basepayload, list);
} else
#endif
{
memcpy (payload, data, size);
gst_rtp_buffer_unmap (&rtp);
- ret = gst_basertppayload_push (basepayload, outbuf);
+ ret = gst_base_rtp_payload_push (basepayload, outbuf);
}
} else {
/* fragmentation Units FU-A */
"recorded %d payload bytes into packet iteration=%d",
limitedSize + 2, ii);
- ret = gst_basertppayload_push (basepayload, outbuf);
+ ret = gst_base_rtp_payload_push (basepayload, outbuf);
if (ret != GST_FLOW_OK)
break;
}
if (rtph264pay->buffer_list) {
/* free iterator and push the whole buffer list at once */
gst_buffer_list_iterator_free (it);
- ret = gst_basertppayload_push_list (basepayload, list);
+ ret = gst_base_rtp_payload_push_list (basepayload, list);
}
#endif
}
if (rtph264pay->sprop_parameter_sets != NULL) {
/* explicitly set profile and sprop, use those */
if (rtph264pay->update_caps) {
- if (!gst_basertppayload_set_outcaps (basepayload,
+ if (!gst_base_rtp_payload_set_outcaps (basepayload,
"sprop-parameter-sets", G_TYPE_STRING,
rtph264pay->sprop_parameter_sets, NULL))
goto caps_rejected;
}
static GstStateChangeReturn
-gst_basertppayload_change_state (GstElement * element,
- GstStateChange transition)
+gst_rtp_h264_pay_change_state (GstElement * element, GstStateChange transition)
{
GstStateChangeReturn ret;
GstRtpH264Pay *rtph264pay = GST_RTP_H264_PAY (element);
if (mode != 20 && mode != 30)
goto wrong_mode;
- gst_basertppayload_set_options (basertppayload, "audio", TRUE, "ILBC", 8000);
+ gst_base_rtp_payload_set_options (basertppayload, "audio", TRUE, "ILBC",
+ 8000);
/* set options for this frame based audio codec */
gst_base_rtp_audio_payload_set_frame_options (basertpaudiopayload,
mode, mode == 30 ? 50 : 38);
mode_str = g_strdup_printf ("%d", mode);
ret =
- gst_basertppayload_set_outcaps (basertppayload, "mode", G_TYPE_STRING,
+ gst_base_rtp_payload_set_outcaps (basertppayload, "mode", G_TYPE_STRING,
mode_str, NULL);
g_free (mode_str);
pay->width = width;
}
- gst_basertppayload_set_options (basepayload, "video", TRUE, "JPEG2000",
+ gst_base_rtp_payload_set_options (basepayload, "video", TRUE, "JPEG2000",
90000);
- res = gst_basertppayload_set_outcaps (basepayload, NULL);
+ res = gst_base_rtp_payload_set_outcaps (basepayload, NULL);
return res;
}
memcpy (header + HEADER_SIZE, &data[offset], data_size);
gst_rtp_buffer_unmap (&rtp);
- ret = gst_basertppayload_push (basepayload, outbuf);
+ ret = gst_base_rtp_payload_push (basepayload, outbuf);
if (ret != GST_FLOW_OK)
goto done;
}
if (pay->buffer_list) {
/* free iterator and push the whole buffer list at once */
gst_buffer_list_iterator_free (it);
- ret = gst_basertppayload_push_list (basepayload, list);
+ ret = gst_base_rtp_payload_push_list (basepayload, list);
}
#endif
}
pay->width = width / 8;
- gst_basertppayload_set_options (basepayload, "video", TRUE, "JPEG", 90000);
- res = gst_basertppayload_set_outcaps (basepayload, NULL);
+ gst_base_rtp_payload_set_options (basepayload, "video", TRUE, "JPEG", 90000);
+ res = gst_base_rtp_payload_set_outcaps (basepayload, NULL);
return res;
/* and add to list */
gst_buffer_list_insert (list, -1, outbuf);
} else {
- ret = gst_basertppayload_push (basepayload, outbuf);
+ ret = gst_base_rtp_payload_push (basepayload, outbuf);
if (ret != GST_FLOW_OK)
break;
}
if (pay->buffer_list) {
/* push the whole buffer list at once */
- ret = gst_basertppayload_push_list (basepayload, list);
+ ret = gst_base_rtp_payload_push_list (basepayload, list);
}
gst_buffer_unmap (buffer, bdata, -1);
{
gboolean res;
- gst_basertppayload_set_options (payload, "video", TRUE, "MP2T", 90000);
- res = gst_basertppayload_set_outcaps (payload, NULL);
+ gst_base_rtp_payload_set_options (payload, "video", TRUE, "MP2T", 90000);
+ res = gst_base_rtp_payload_set_outcaps (payload, NULL);
return res;
}
GST_DEBUG_OBJECT (rtpmp2tpay, "pushing buffer of size %d",
gst_buffer_get_size (outbuf));
- ret = gst_basertppayload_push (GST_BASE_RTP_PAYLOAD (rtpmp2tpay), outbuf);
+ ret = gst_base_rtp_payload_push (GST_BASE_RTP_PAYLOAD (rtpmp2tpay), outbuf);
/* flush the adapter content */
gst_adapter_flush (rtpmp2tpay->adapter, avail);
/* if this buffer is going to overflow the packet, flush what we
* have. */
- if (gst_basertppayload_is_filled (basepayload,
+ if (gst_base_rtp_payload_is_filled (basepayload,
packet_len, rtpmp2tpay->duration + duration)) {
ret = gst_rtp_mp2t_pay_flush (rtpmp2tpay);
rtpmp2tpay->first_ts = timestamp;
gst_value_set_buffer (&v, rtpmp4apay->config);
config = gst_value_serialize (&v);
- res = gst_basertppayload_set_outcaps (GST_BASE_RTP_PAYLOAD (rtpmp4apay),
+ res = gst_base_rtp_payload_set_outcaps (GST_BASE_RTP_PAYLOAD (rtpmp4apay),
"cpresent", G_TYPE_STRING, "0", "config", G_TYPE_STRING, config, NULL);
g_value_unset (&v);
GST_WARNING_OBJECT (payload, "Need framed AAC data as input!");
}
- gst_basertppayload_set_options (payload, "audio", TRUE, "MP4A-LATM",
+ gst_base_rtp_payload_set_options (payload, "audio", TRUE, "MP4A-LATM",
rtpmp4apay->rate);
res = gst_rtp_mp4a_pay_new_caps (rtpmp4apay);
/* copy incomming timestamp (if any) to outgoing buffers */
GST_BUFFER_TIMESTAMP (outbuf) = timestamp;
- ret = gst_basertppayload_push (GST_BASE_RTP_PAYLOAD (rtpmp4apay), outbuf);
+ ret = gst_base_rtp_payload_push (GST_BASE_RTP_PAYLOAD (rtpmp4apay), outbuf);
fragmented = TRUE;
}
/* hmm, silly */
if (rtpmp4gpay->params) {
- res = gst_basertppayload_set_outcaps (GST_BASE_RTP_PAYLOAD (rtpmp4gpay),
+ res = gst_base_rtp_payload_set_outcaps (GST_BASE_RTP_PAYLOAD (rtpmp4gpay),
"encoding-params", G_TYPE_STRING, rtpmp4gpay->params, MP4GCAPS);
} else {
- res = gst_basertppayload_set_outcaps (GST_BASE_RTP_PAYLOAD (rtpmp4gpay),
+ res = gst_base_rtp_payload_set_outcaps (GST_BASE_RTP_PAYLOAD (rtpmp4gpay),
MP4GCAPS);
}
if (media_type == NULL)
goto config_failed;
- gst_basertppayload_set_options (payload, media_type, TRUE, "MPEG4-GENERIC",
+ gst_base_rtp_payload_set_options (payload, media_type, TRUE, "MPEG4-GENERIC",
rtpmp4gpay->rate);
res = gst_rtp_mp4g_pay_new_caps (rtpmp4gpay);
rtpmp4gpay->offset += rtpmp4gpay->frame_len;
}
- ret = gst_basertppayload_push (GST_BASE_RTP_PAYLOAD (rtpmp4gpay), outbuf);
+ ret = gst_base_rtp_payload_push (GST_BASE_RTP_PAYLOAD (rtpmp4gpay), outbuf);
avail -= payload_len;
}
gst_value_set_buffer (&v, rtpmp4vpay->config);
config = gst_value_serialize (&v);
- res = gst_basertppayload_set_outcaps (GST_BASE_RTP_PAYLOAD (rtpmp4vpay),
+ res = gst_base_rtp_payload_set_outcaps (GST_BASE_RTP_PAYLOAD (rtpmp4vpay),
"profile-level-id", G_TYPE_STRING, profile,
"config", G_TYPE_STRING, config, NULL);
rtpmp4vpay = GST_RTP_MP4V_PAY (payload);
- gst_basertppayload_set_options (payload, "video", TRUE, "MP4V-ES",
+ gst_base_rtp_payload_set_options (payload, "video", TRUE, "MP4V-ES",
rtpmp4vpay->rate);
res = TRUE;
/* add to list */
gst_buffer_list_insert (list, -1, outbuf);
} else {
- ret = gst_basertppayload_push (GST_BASE_RTP_PAYLOAD (rtpmp4vpay), outbuf);
+ ret =
+ gst_base_rtp_payload_push (GST_BASE_RTP_PAYLOAD (rtpmp4vpay), outbuf);
}
}
if (rtpmp4vpay->buffer_list) {
/* push the whole buffer list at once */
ret =
- gst_basertppayload_push_list (GST_BASE_RTP_PAYLOAD (rtpmp4vpay), list);
+ gst_base_rtp_payload_push_list (GST_BASE_RTP_PAYLOAD (rtpmp4vpay),
+ list);
}
return ret;
/* get packet length of data and see if we exceeded MTU. */
packet_len = gst_rtp_buffer_calc_packet_len (avail + size, 0, 0);
- if (gst_basertppayload_is_filled (basepayload,
+ if (gst_base_rtp_payload_is_filled (basepayload,
packet_len, rtpmp4vpay->duration + duration)) {
ret = gst_rtp_mp4v_pay_flush (rtpmp4vpay);
rtpmp4vpay->first_timestamp = timestamp;
{
gboolean res;
- gst_basertppayload_set_options (payload, "audio", TRUE, "MPA", 90000);
- res = gst_basertppayload_set_outcaps (payload, NULL);
+ gst_base_rtp_payload_set_options (payload, "audio", TRUE, "MPA", 90000);
+ res = gst_base_rtp_payload_set_outcaps (payload, NULL);
return res;
}
GST_BUFFER_TIMESTAMP (outbuf) = rtpmpapay->first_ts;
GST_BUFFER_DURATION (outbuf) = rtpmpapay->duration;
- ret = gst_basertppayload_push (GST_BASE_RTP_PAYLOAD (rtpmpapay), outbuf);
+ ret = gst_base_rtp_payload_push (GST_BASE_RTP_PAYLOAD (rtpmpapay), outbuf);
}
return ret;
/* if this buffer is going to overflow the packet, flush what we
* have. */
- if (gst_basertppayload_is_filled (basepayload,
+ if (gst_base_rtp_payload_is_filled (basepayload,
packet_len, rtpmpapay->duration + duration)) {
ret = gst_rtp_mpa_pay_flush (rtpmpapay);
avail = 0;
static gboolean
gst_rtp_mpv_pay_setcaps (GstBaseRTPPayload * payload, GstCaps * caps)
{
- gst_basertppayload_set_options (payload, "video", FALSE, "MPV", 90000);
- return gst_basertppayload_set_outcaps (payload, NULL);
+ gst_base_rtp_payload_set_options (payload, "video", FALSE, "MPV", 90000);
+ return gst_base_rtp_payload_set_outcaps (payload, NULL);
}
static gboolean
GST_BUFFER_TIMESTAMP (outbuf) = rtpmpvpay->first_ts;
- ret = gst_basertppayload_push (GST_BASE_RTP_PAYLOAD (rtpmpvpay), outbuf);
+ ret = gst_base_rtp_payload_push (GST_BASE_RTP_PAYLOAD (rtpmpvpay), outbuf);
}
return ret;
GST_LOG_OBJECT (rtpmpvpay, "available %d, rtp packet length %d", avail,
packet_len);
- if (gst_basertppayload_is_filled (basepayload,
+ if (gst_base_rtp_payload_is_filled (basepayload,
packet_len, rtpmpvpay->duration)) {
ret = gst_rtp_mpv_pay_flush (rtpmpvpay);
} else {
payload->pt = GST_RTP_PAYLOAD_PCMA;
- gst_basertppayload_set_options (payload, "audio", FALSE, "PCMA", 8000);
- res = gst_basertppayload_set_outcaps (payload, NULL);
+ gst_base_rtp_payload_set_options (payload, "audio", FALSE, "PCMA", 8000);
+ res = gst_base_rtp_payload_set_outcaps (payload, NULL);
return res;
}
payload->pt = GST_RTP_PAYLOAD_PCMU;
- gst_basertppayload_set_options (payload, "audio", FALSE, "PCMU", 8000);
- res = gst_basertppayload_set_outcaps (payload, NULL);
+ gst_base_rtp_payload_set_options (payload, "audio", FALSE, "PCMU", 8000);
+ res = gst_base_rtp_payload_set_outcaps (payload, NULL);
return res;
}
if (g_ascii_strcasecmp ("audio/x-siren", payload_name))
goto wrong_caps;
- gst_basertppayload_set_options (basertppayload, "audio", TRUE, "SIREN",
+ gst_base_rtp_payload_set_options (basertppayload, "audio", TRUE, "SIREN",
16000);
/* set options for this frame based audio codec */
gst_base_rtp_audio_payload_set_frame_options (basertpaudiopayload, 20, 40);
- return gst_basertppayload_set_outcaps (basertppayload, NULL);
+ return gst_base_rtp_payload_set_outcaps (basertppayload, NULL);
/* ERRORS */
wrong_dct:
payload = GST_BASE_RTP_PAYLOAD (rtpspeexpay);
- gst_basertppayload_set_options (payload, "audio", FALSE, "SPEEX", rate);
+ gst_base_rtp_payload_set_options (payload, "audio", FALSE, "SPEEX", rate);
cstr = g_strdup_printf ("%d", nb_channels);
- res = gst_basertppayload_set_outcaps (payload, "encoding-params",
+ res = gst_base_rtp_payload_set_outcaps (payload, "encoding-params",
G_TYPE_STRING, cstr, NULL);
g_free (cstr);
gst_rtp_buffer_unmap (&rtp);
- ret = gst_basertppayload_push (basepayload, outbuf);
+ ret = gst_base_rtp_payload_push (basepayload, outbuf);
done:
gst_buffer_unmap (buffer, data, -1);
GstFlowReturn ret;
gint payload_len;
guint8 *payload, *to_free = NULL;
- guint32 timestamp;
guint32 header, ident;
guint8 F, TDT, packets;
GstRTPBuffer rtp;
* .. theora data |
* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+*
*/
- timestamp = gst_rtp_buffer_get_timestamp (&rtp);
-
while (payload_len >= 2) {
guint16 length;
payload += length;
payload_len -= length;
- if (timestamp != -1)
- /* push with timestamp of the last packet, which is the same timestamp that
- * should apply to the first assembled packet. */
- ret = gst_base_rtp_depayload_push_ts (depayload, timestamp, outbuf);
- else
- ret = gst_base_rtp_depayload_push (depayload, outbuf);
-
+ ret = gst_base_rtp_depayload_push (depayload, outbuf);
if (ret != GST_FLOW_OK)
break;
-
- /* make sure we don't set a timestamp on next buffers */
- timestamp = -1;
}
g_free (to_free);
/* push, this gives away our ref to the packet, so clear it. */
ret =
- gst_basertppayload_push (GST_BASE_RTP_PAYLOAD (rtptheorapay),
+ gst_base_rtp_payload_push (GST_BASE_RTP_PAYLOAD (rtptheorapay),
rtptheorapay->packet);
rtptheorapay->packet = NULL;
/* configure payloader settings */
wstr = g_strdup_printf ("%d", rtptheorapay->width);
hstr = g_strdup_printf ("%d", rtptheorapay->height);
- gst_basertppayload_set_options (basepayload, "video", TRUE, "THEORA", 90000);
- res = gst_basertppayload_set_outcaps (basepayload,
- "sampling", G_TYPE_STRING, "YCbCr-4:2:0",
- "width", G_TYPE_STRING, wstr,
- "height", G_TYPE_STRING, hstr,
- "configuration", G_TYPE_STRING, configuration,
- "delivery-method", G_TYPE_STRING, "inline",
+ gst_base_rtp_payload_set_options (basepayload, "video", TRUE, "THEORA",
+ 90000);
+ res =
+ gst_base_rtp_payload_set_outcaps (basepayload, "sampling", G_TYPE_STRING,
+ "YCbCr-4:2:0", "width", G_TYPE_STRING, wstr, "height", G_TYPE_STRING,
+ hstr, "configuration", G_TYPE_STRING, configuration, "delivery-method",
+ G_TYPE_STRING, "inline",
/* don't set the other defaults
*/
NULL);
packet_len = gst_rtp_buffer_calc_packet_len (newsize, 0, 0);
/* check buffer filled against length and max latency */
- flush = gst_basertppayload_is_filled (GST_BASE_RTP_PAYLOAD (rtptheorapay),
+ flush = gst_base_rtp_payload_is_filled (GST_BASE_RTP_PAYLOAD (rtptheorapay),
packet_len, newduration);
/* we can store up to 15 theora packets in one RTP packet. */
flush |= (rtptheorapay->payload_pkts == 15);
GstFlowReturn ret;
gint payload_len;
guint8 *payload, *to_free = NULL;
- guint32 timestamp;
guint32 header, ident;
guint8 F, VDT, packets;
GstRTPBuffer rtp;
* .. vorbis data |
* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+*
*/
- timestamp = gst_rtp_buffer_get_timestamp (&rtp);
-
while (payload_len > 2) {
guint16 length;
payload += length;
payload_len -= length;
- if (timestamp != -1)
- /* push with timestamp of the last packet, which is the same timestamp that
- * should apply to the first assembled packet. */
- ret = gst_base_rtp_depayload_push_ts (depayload, timestamp, outbuf);
- else
- ret = gst_base_rtp_depayload_push (depayload, outbuf);
-
+ ret = gst_base_rtp_depayload_push (depayload, outbuf);
if (ret != GST_FLOW_OK)
break;
-
- /* make sure we don't set a timestamp on next buffers */
- timestamp = -1;
}
g_free (to_free);
/* push, this gives away our ref to the packet, so clear it. */
ret =
- gst_basertppayload_push (GST_BASE_RTP_PAYLOAD (rtpvorbispay),
+ gst_base_rtp_payload_push (GST_BASE_RTP_PAYLOAD (rtpvorbispay),
rtpvorbispay->packet);
rtpvorbispay->packet = NULL;
/* configure payloader settings */
cstr = g_strdup_printf ("%d", rtpvorbispay->channels);
- gst_basertppayload_set_options (basepayload, "audio", TRUE, "VORBIS",
+ gst_base_rtp_payload_set_options (basepayload, "audio", TRUE, "VORBIS",
rtpvorbispay->rate);
res =
- gst_basertppayload_set_outcaps (basepayload, "encoding-params",
+ gst_base_rtp_payload_set_outcaps (basepayload, "encoding-params",
G_TYPE_STRING, cstr, "configuration", G_TYPE_STRING, configuration, NULL);
g_free (cstr);
g_free (configuration);
packet_len = gst_rtp_buffer_calc_packet_len (newsize, 0, 0);
/* check buffer filled against length and max latency */
- flush = gst_basertppayload_is_filled (basepayload, packet_len, newduration);
+ flush = gst_base_rtp_payload_is_filled (basepayload, packet_len, newduration);
/* we can store up to 15 vorbis packets in one RTP packet. */
flush |= (rtpvorbispay->payload_pkts == 15);
/* flush if we have a new VDT */
GST_LOG_OBJECT (depayload, "new frame with timestamp %u", timestamp);
/* new timestamp, flush old buffer and create new output buffer */
if (rtpvrawdepay->outbuf) {
- gst_base_rtp_depayload_push_ts (depayload, rtpvrawdepay->timestamp,
- rtpvrawdepay->outbuf);
+ gst_base_rtp_depayload_push (depayload, rtpvrawdepay->outbuf);
rtpvrawdepay->outbuf = NULL;
}
if (gst_rtp_buffer_get_marker (&rtp)) {
GST_LOG_OBJECT (depayload, "marker, flushing frame");
if (rtpvrawdepay->outbuf) {
- gst_base_rtp_depayload_push_ts (depayload, timestamp,
- rtpvrawdepay->outbuf);
+ gst_base_rtp_depayload_push (depayload, rtpvrawdepay->outbuf);
rtpvrawdepay->outbuf = NULL;
}
rtpvrawdepay->timestamp = -1;
wstr = g_strdup_printf ("%d", GST_VIDEO_INFO_WIDTH (&info));
hstr = g_strdup_printf ("%d", GST_VIDEO_INFO_HEIGHT (&info));
- gst_basertppayload_set_options (payload, "video", TRUE, "RAW", 90000);
+ gst_base_rtp_payload_set_options (payload, "video", TRUE, "RAW", 90000);
if (info.flags & GST_VIDEO_FLAG_INTERLACED) {
- res = gst_basertppayload_set_outcaps (payload, "sampling", G_TYPE_STRING,
+ res = gst_base_rtp_payload_set_outcaps (payload, "sampling", G_TYPE_STRING,
samplingstr, "depth", G_TYPE_STRING, depthstr, "width", G_TYPE_STRING,
wstr, "height", G_TYPE_STRING, hstr, "colorimetry", G_TYPE_STRING,
colorimetrystr, "interlace", G_TYPE_STRING, "true", NULL);
} else {
- res = gst_basertppayload_set_outcaps (payload, "sampling", G_TYPE_STRING,
+ res = gst_base_rtp_payload_set_outcaps (payload, "sampling", G_TYPE_STRING,
samplingstr, "depth", G_TYPE_STRING, depthstr, "width", G_TYPE_STRING,
wstr, "height", G_TYPE_STRING, hstr, "colorimetry", G_TYPE_STRING,
colorimetrystr, NULL);
}
/* push buffer */
- ret = gst_basertppayload_push (payload, out);
+ ret = gst_base_rtp_payload_push (payload, out);
}
}