static gboolean
gst_dv_pay_negotiate (GstRTPDVPay * rtpdvpay, guint8 * data, guint size)
{
const gchar *encode, *media;
gboolean audio_bundled, res;
if ((data[3] & 0x80) == 0) { /* DSF flag */
/* it's an NTSC format */
if ((data[80 * 5 + 48 + 3] & 0x4) && (data[80 * 5 + 48] == 0x60)) { /* 4:2:2 sampling */
/* NTSC 50Mbps */
encode = "314M-25/525-60";
} else { /* 4:1:1 sampling */
/* NTSC 25Mbps */
encode = "SD-VCR/525-60";
}
} else {
/* it's a PAL format */
if ((data[80 * 5 + 48 + 3] & 0x4) && (data[80 * 5 + 48] == 0x60)) { /* 4:2:2 sampling */
/* PAL 50Mbps */
encode = "314M-50/625-50";
} else if ((data[5] & 0x07) == 0) { /* APT flag */
/* PAL 25Mbps 4:2:0 */
encode = "SD-VCR/625-50";
} else
/* PAL 25Mbps 4:1:1 */
encode = "314M-25/625-50";
}
media = "video";
audio_bundled = FALSE;
switch (rtpdvpay->mode) {
case GST_DV_PAY_MODE_AUDIO:
media = "audio";
break;
case GST_DV_PAY_MODE_BUNDLED:
audio_bundled = TRUE;
break;
default:
break;
}
gst_basertppayload_set_options (GST_BASE_RTP_PAYLOAD (rtpdvpay), media, TRUE,
"DV", 90000);
if (audio_bundled) {
res = gst_basertppayload_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),
"encode", G_TYPE_STRING, encode, NULL);
}
return res;
}
static GstCaps *
gst_basertppayload_sink_getcaps (GstPad * pad)
{
GstBaseRTPPayload *basertppayload;
GstBaseRTPPayloadClass *basertppayload_class;
GstCaps *caps = NULL;
GST_DEBUG_OBJECT (pad, "getting caps");
basertppayload = GST_BASE_RTP_PAYLOAD (gst_pad_get_parent (pad));
basertppayload_class = GST_BASE_RTP_PAYLOAD_GET_CLASS (basertppayload);
if (basertppayload_class->get_caps)
caps = basertppayload_class->get_caps (basertppayload, pad);
if (!caps) {
caps = GST_PAD_TEMPLATE_CAPS (GST_PAD_PAD_TEMPLATE (pad));
GST_DEBUG_OBJECT (pad,
"using pad template %p with caps %p %" GST_PTR_FORMAT,
GST_PAD_PAD_TEMPLATE (pad), caps, caps);
caps = gst_caps_ref (caps);
}
gst_object_unref (basertppayload);
return caps;
}
/**
* gst_base_rtp_audio_payload_push:
* @baseaudiopayload: a #GstBaseRTPPayload
* @data: data to set as payload
* @payload_len: length of payload
* @timestamp: a #GstClockTime
*
* Create an RTP buffer and store @payload_len bytes of @data as the
* payload. Set the timestamp on the new buffer to @timestamp before pushing
* the buffer downstream.
*
* Returns: a #GstFlowReturn
*
* Since: 0.10.13
*/
GstFlowReturn
gst_base_rtp_audio_payload_push (GstBaseRTPAudioPayload * baseaudiopayload,
const guint8 * data, guint payload_len, GstClockTime timestamp)
{
GstBaseRTPPayload *basepayload;
GstBuffer *outbuf;
guint8 *payload;
GstFlowReturn ret;
basepayload = GST_BASE_RTP_PAYLOAD (baseaudiopayload);
GST_DEBUG_OBJECT (baseaudiopayload, "Pushing %d bytes ts %" GST_TIME_FORMAT,
payload_len, GST_TIME_ARGS (timestamp));
/* create buffer to hold the payload */
outbuf = gst_rtp_buffer_new_allocate (payload_len, 0, 0);
/* copy payload */
payload = gst_rtp_buffer_get_payload (outbuf);
memcpy (payload, data, payload_len);
/* set metadata */
gst_base_rtp_audio_payload_set_meta (baseaudiopayload, outbuf, payload_len,
timestamp);
ret = gst_basertppayload_push (basepayload, outbuf);
return ret;
}
static void
gst_rtp_speex_pay_init (GstRtpSPEEXPay * rtpspeexpay,
GstRtpSPEEXPayClass * klass)
{
GST_BASE_RTP_PAYLOAD (rtpspeexpay)->clock_rate = 8000;
GST_BASE_RTP_PAYLOAD_PT (rtpspeexpay) = 110; /* Create String */
}
static GstFlowReturn
gst_basertppayload_chain (GstPad * pad, GstBuffer * buffer)
{
GstBaseRTPPayload *basertppayload;
GstBaseRTPPayloadClass *basertppayload_class;
GstFlowReturn ret;
basertppayload = GST_BASE_RTP_PAYLOAD (gst_pad_get_parent (pad));
basertppayload_class = GST_BASE_RTP_PAYLOAD_GET_CLASS (basertppayload);
if (!basertppayload_class->handle_buffer)
goto no_function;
ret = basertppayload_class->handle_buffer (basertppayload, buffer);
gst_object_unref (basertppayload);
return ret;
/* ERRORS */
no_function:
{
GST_ELEMENT_ERROR (basertppayload, STREAM, NOT_IMPLEMENTED, (NULL),
("subclass did not implement handle_buffer function"));
gst_object_unref (basertppayload);
gst_buffer_unref (buffer);
return GST_FLOW_ERROR;
}
}
static GstFlowReturn
gst_rtp_g723_pay_flush (GstRTPG723Pay * pay)
{
GstBuffer *outbuf;
GstFlowReturn ret;
guint8 *payload;
guint avail;
avail = gst_adapter_available (pay->adapter);
outbuf = gst_rtp_buffer_new_allocate (avail, 0, 0);
payload = gst_rtp_buffer_get_payload (outbuf);
GST_BUFFER_TIMESTAMP (outbuf) = pay->timestamp;
GST_BUFFER_DURATION (outbuf) = pay->duration;
/* copy G723 data as payload */
gst_adapter_copy (pay->adapter, payload, 0, avail);
/* flush bytes from adapter */
gst_adapter_flush (pay->adapter, avail);
pay->timestamp = GST_CLOCK_TIME_NONE;
pay->duration = 0;
/* set discont and marker */
if (pay->discont) {
GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
gst_rtp_buffer_set_marker (outbuf, TRUE);
pay->discont = FALSE;
}
ret = gst_basertppayload_push (GST_BASE_RTP_PAYLOAD (pay), outbuf);
return ret;
}
static GstFlowReturn
gst_rtp_mp2t_pay_flush (GstRTPMP2TPay * rtpmp2tpay)
{
guint avail;
guint8 *payload;
GstFlowReturn ret;
GstBuffer *outbuf;
avail = gst_adapter_available (rtpmp2tpay->adapter);
if (avail == 0)
return GST_FLOW_OK;
outbuf = gst_rtp_buffer_new_allocate (avail, 0, 0);
/* get payload */
payload = gst_rtp_buffer_get_payload (outbuf);
/* copy stuff from adapter to payload */
gst_adapter_copy (rtpmp2tpay->adapter, payload, 0, avail);
GST_BUFFER_TIMESTAMP (outbuf) = rtpmp2tpay->first_ts;
GST_BUFFER_DURATION (outbuf) = rtpmp2tpay->duration;
GST_DEBUG_OBJECT (rtpmp2tpay, "pushing buffer of size %d",
GST_BUFFER_SIZE (outbuf));
ret = gst_basertppayload_push (GST_BASE_RTP_PAYLOAD (rtpmp2tpay), outbuf);
/* flush the adapter content */
gst_adapter_flush (rtpmp2tpay->adapter, avail);
return ret;
}
static GstFlowReturn
gst_rtp_mpv_pay_flush (GstRTPMPVPay * rtpmpvpay, GstClockTime timestamp,
GstClockTime duration)
{
GstBuffer *outbuf;
GstFlowReturn ret;
guint avail;
guint8 *payload;
gint packet_size;
gint payload_size;
avail = gst_adapter_available (rtpmpvpay->adapter);
packet_size = gst_rtp_buffer_calc_packet_len (4 + avail, 0, 0);
/* check for the maximum size of the rtp buffer */
if (packet_size > GST_BASE_RTP_PAYLOAD_MTU (rtpmpvpay)) {
payload_size =
GST_BASE_RTP_PAYLOAD_MTU (rtpmpvpay) -
gst_rtp_buffer_calc_packet_len (4, 0, 0);
} else {
payload_size = avail;
}
outbuf = gst_rtp_buffer_new_allocate (4 + payload_size, 0, 0);
/* enable MPEG Video-specific header
*
* 0 1 2 3
* 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
* | MBZ |T| TR | |N|S|B|E| P | | BFC | | FFC |
* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
* AN FBV FFV
*/
payload = gst_rtp_buffer_get_payload (outbuf);
/* fill in the MPEG Video-specific header */
memset (payload, 0x0, 4);
/* copy stuff from adapter to payload */
gst_adapter_copy (rtpmpvpay->adapter, payload + 4, 0, payload_size);
GST_BUFFER_TIMESTAMP (outbuf) = rtpmpvpay->first_ts;
GST_BUFFER_DURATION (outbuf) = rtpmpvpay->duration;
GST_DEBUG_OBJECT (rtpmpvpay, "pushing buffer of size %d",
GST_BUFFER_SIZE (outbuf));
ret = gst_basertppayload_push (GST_BASE_RTP_PAYLOAD (rtpmpvpay), outbuf);
gst_adapter_flush (rtpmpvpay->adapter, payload_size);
/* update the timestamp and duration */
rtpmpvpay->first_ts = timestamp;
rtpmpvpay->duration = duration;
/* check if there is enough data for another rtp buffer */
avail = gst_adapter_available (rtpmpvpay->adapter);
packet_size = gst_rtp_buffer_calc_packet_len (4 + avail, 0, 0);
if (packet_size >= GST_BASE_RTP_PAYLOAD_MTU (rtpmpvpay) && ret == GST_FLOW_OK) {
GST_DEBUG_OBJECT (rtpmpvpay, "Have enough data for another rtp packet");
ret = gst_rtp_mpv_pay_flush (rtpmpvpay, timestamp, duration);
}
return ret;
}
static void
gst_rtp_mpv_pay_init (GstRTPMPVPay * rtpmpvpay, GstRTPMPVPayClass * klass)
{
GST_BASE_RTP_PAYLOAD (rtpmpvpay)->clock_rate = 90000;
GST_BASE_RTP_PAYLOAD_PT (rtpmpvpay) = GST_RTP_PAYLOAD_MPV;
rtpmpvpay->adapter = gst_adapter_new ();
}
static GstFlowReturn
gst_rtp_mpv_pay_flush (GstRTPMPVPay * rtpmpvpay)
{
GstBuffer *outbuf;
GstFlowReturn ret;
guint avail;
guint8 *payload;
avail = gst_adapter_available (rtpmpvpay->adapter);
ret = GST_FLOW_OK;
while (avail > 0) {
guint towrite;
guint packet_len;
guint payload_len;
packet_len = gst_rtp_buffer_calc_packet_len (avail, 4, 0);
towrite = MIN (packet_len, GST_BASE_RTP_PAYLOAD_MTU (rtpmpvpay));
payload_len = gst_rtp_buffer_calc_payload_len (towrite, 4, 0);
outbuf = gst_rtp_buffer_new_allocate (payload_len, 4, 0);
payload = gst_rtp_buffer_get_payload (outbuf);
/* enable MPEG Video-specific header
*
* 0 1 2 3
* 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
* | MBZ |T| TR | |N|S|B|E| P | | BFC | | FFC |
* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
* AN FBV FFV
*/
/* fill in the MPEG Video-specific header
* data is set to 0x0 here
*/
memset (payload, 0x0, 4);
gst_adapter_copy (rtpmpvpay->adapter, payload + 4, 0, payload_len);
gst_adapter_flush (rtpmpvpay->adapter, payload_len);
avail -= payload_len;
gst_rtp_buffer_set_marker (outbuf, avail == 0);
GST_BUFFER_TIMESTAMP (outbuf) = rtpmpvpay->first_ts;
ret = gst_basertppayload_push (GST_BASE_RTP_PAYLOAD (rtpmpvpay), outbuf);
}
return ret;
}
static GstFlowReturn
gst_rtp_celt_pay_flush_queued (GstRtpCELTPay * rtpceltpay)
{
GstFlowReturn ret;
GstBuffer *buf, *outbuf;
guint8 *payload, *spayload;
guint payload_len;
GstClockTime duration;
payload_len = rtpceltpay->bytes + rtpceltpay->sbytes;
duration = rtpceltpay->qduration;
GST_DEBUG_OBJECT (rtpceltpay, "flushing out %u, duration %" GST_TIME_FORMAT,
payload_len, GST_TIME_ARGS (rtpceltpay->qduration));
/* get a big enough packet for the sizes + payloads */
outbuf = gst_rtp_buffer_new_allocate (payload_len, 0, 0);
GST_BUFFER_DURATION (outbuf) = duration;
/* point to the payload for size headers and data */
spayload = gst_rtp_buffer_get_payload (outbuf);
payload = spayload + rtpceltpay->sbytes;
while ((buf = g_queue_pop_head (rtpceltpay->queue))) {
guint size;
/* copy first timestamp to output */
if (GST_BUFFER_TIMESTAMP (outbuf) == -1)
GST_BUFFER_TIMESTAMP (outbuf) = GST_BUFFER_TIMESTAMP (buf);
/* write the size to the header */
size = GST_BUFFER_SIZE (buf);
while (size > 0xff) {
*spayload++ = 0xff;
size -= 0xff;
}
*spayload++ = size;
size = GST_BUFFER_SIZE (buf);
/* copy payload */
memcpy (payload, GST_BUFFER_DATA (buf), size);
payload += size;
gst_buffer_unref (buf);
}
/* we consumed it all */
rtpceltpay->bytes = 0;
rtpceltpay->sbytes = 0;
rtpceltpay->qduration = 0;
ret = gst_basertppayload_push (GST_BASE_RTP_PAYLOAD (rtpceltpay), outbuf);
return ret;
}
static void
gst_rtp_g723_pay_init (GstRTPG723Pay * pay, GstRTPG723PayClass * klass)
{
GstBaseRTPPayload *payload = GST_BASE_RTP_PAYLOAD (pay);
pay->adapter = gst_adapter_new ();
payload->pt = GST_RTP_PAYLOAD_G723;
gst_basertppayload_set_options (payload, "audio", FALSE, "G723", 8000);
}
static guint32
gst_base_rtp_audio_payload_frame_bytes_to_rtptime (GstBaseRTPAudioPayload *
payload, guint64 bytes)
{
guint64 time;
time = (bytes / payload->frame_size) * (payload->priv->frame_duration_ns);
return gst_util_uint64_scale_int (time,
GST_BASE_RTP_PAYLOAD (payload)->clock_rate, GST_SECOND);
}
static void
gst_basertppayload_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstBaseRTPPayload *basertppayload;
GstBaseRTPPayloadPrivate *priv;
basertppayload = GST_BASE_RTP_PAYLOAD (object);
priv = basertppayload->priv;
switch (prop_id) {
case PROP_MTU:
g_value_set_uint (value, basertppayload->mtu);
break;
case PROP_PT:
g_value_set_uint (value, basertppayload->pt);
break;
case PROP_SSRC:
if (priv->ssrc_random)
g_value_set_uint (value, -1);
else
g_value_set_uint (value, basertppayload->ssrc);
break;
case PROP_TIMESTAMP_OFFSET:
if (priv->ts_offset_random)
g_value_set_uint (value, -1);
else
g_value_set_uint (value, (guint32) basertppayload->ts_offset);
break;
case PROP_SEQNUM_OFFSET:
if (priv->seqnum_offset_random)
g_value_set_int (value, -1);
else
g_value_set_int (value, (guint16) basertppayload->seqnum_offset);
break;
case PROP_MAX_PTIME:
g_value_set_int64 (value, basertppayload->max_ptime);
break;
case PROP_MIN_PTIME:
g_value_set_int64 (value, basertppayload->min_ptime);
break;
case PROP_TIMESTAMP:
g_value_set_uint (value, basertppayload->timestamp);
break;
case PROP_SEQNUM:
g_value_set_uint (value, basertppayload->seqnum);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static GstFlowReturn
gst_rtp_mp4v_pay_flush (GstRtpMP4VPay * rtpmp4vpay)
{
guint avail;
GstBuffer *outbuf;
GstFlowReturn ret;
/* the data available in the adapter is either smaller
* than the MTU or bigger. In the case it is smaller, the complete
* adapter contents can be put in one packet. In the case the
* adapter has more than one MTU, we need to split the MP4V data
* over multiple packets. */
avail = gst_adapter_available (rtpmp4vpay->adapter);
ret = GST_FLOW_OK;
while (avail > 0) {
guint towrite;
guint8 *payload;
guint payload_len;
guint packet_len;
/* this will be the total lenght of the packet */
packet_len = gst_rtp_buffer_calc_packet_len (avail, 0, 0);
/* fill one MTU or all available bytes */
towrite = MIN (packet_len, GST_BASE_RTP_PAYLOAD_MTU (rtpmp4vpay));
/* this is the payload length */
payload_len = gst_rtp_buffer_calc_payload_len (towrite, 0, 0);
/* create buffer to hold the payload */
outbuf = gst_rtp_buffer_new_allocate (payload_len, 0, 0);
/* copy payload */
payload = gst_rtp_buffer_get_payload (outbuf);
gst_adapter_copy (rtpmp4vpay->adapter, payload, 0, payload_len);
gst_adapter_flush (rtpmp4vpay->adapter, payload_len);
avail -= payload_len;
gst_rtp_buffer_set_marker (outbuf, avail == 0);
GST_BUFFER_TIMESTAMP (outbuf) = rtpmp4vpay->first_timestamp;
ret = gst_basertppayload_push (GST_BASE_RTP_PAYLOAD (rtpmp4vpay), outbuf);
}
return ret;
}
static GstStateChangeReturn
gst_basertppayload_change_state (GstElement * element,
GstStateChange transition)
{
GstBaseRTPPayload *basertppayload;
GstBaseRTPPayloadPrivate *priv;
GstStateChangeReturn ret;
basertppayload = GST_BASE_RTP_PAYLOAD (element);
priv = basertppayload->priv;
switch (transition) {
case GST_STATE_CHANGE_NULL_TO_READY:
break;
case GST_STATE_CHANGE_READY_TO_PAUSED:
gst_segment_init (&basertppayload->segment, GST_FORMAT_UNDEFINED);
if (priv->seqnum_offset_random)
basertppayload->seqnum_base =
g_rand_int_range (basertppayload->seq_rand, 0, G_MAXUINT16);
else
basertppayload->seqnum_base = basertppayload->seqnum_offset;
priv->next_seqnum = basertppayload->seqnum_base;
basertppayload->seqnum = basertppayload->seqnum_base;
if (priv->ssrc_random)
basertppayload->current_ssrc = g_rand_int (basertppayload->ssrc_rand);
else
basertppayload->current_ssrc = basertppayload->ssrc;
if (priv->ts_offset_random)
basertppayload->ts_base = g_rand_int (basertppayload->ts_rand);
else
basertppayload->ts_base = basertppayload->ts_offset;
basertppayload->timestamp = basertppayload->ts_base;
break;
default:
break;
}
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
switch (transition) {
case GST_STATE_CHANGE_READY_TO_NULL:
break;
default:
break;
}
return ret;
}
static void
gst_rtp_pcma_pay_init (GstRtpPmcaPay * rtppcmapay, GstRtpPmcaPayClass * klass)
{
GstBaseRTPAudioPayload *basertpaudiopayload;
basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (rtppcmapay);
GST_BASE_RTP_PAYLOAD (rtppcmapay)->clock_rate = 8000;
/* tell basertpaudiopayload that this is a sample based codec */
gst_base_rtp_audio_payload_set_sample_based (basertpaudiopayload);
/* octet-per-sample is 1 for PCM */
gst_base_rtp_audio_payload_set_sample_options (basertpaudiopayload, 1);
}
static guint64
gst_base_rtp_audio_payload_sample_time_to_bytes (GstBaseRTPAudioPayload *
payload, guint64 time)
{
guint64 samples;
samples = gst_util_uint64_scale_int (time,
GST_BASE_RTP_PAYLOAD (payload)->clock_rate, GST_SECOND);
/* avoid multiplication when we can */
if (G_LIKELY (payload->sample_size != 8))
return gst_util_uint64_scale_int (samples, payload->sample_size, 8);
else
return samples;
}
static gboolean
gst_rtp_mp4g_pay_new_caps (GstRtpMP4GPay * rtpmp4gpay)
{
gchar *config;
GValue v = { 0 };
gboolean res;
#define MP4GCAPS \
"streamtype", G_TYPE_STRING, rtpmp4gpay->streamtype, \
"profile-level-id", G_TYPE_STRING, rtpmp4gpay->profile, \
"mode", G_TYPE_STRING, rtpmp4gpay->mode, \
"config", G_TYPE_STRING, config, \
"sizelength", G_TYPE_STRING, "13", \
"indexlength", G_TYPE_STRING, "3", \
"indexdeltalength", G_TYPE_STRING, "3", \
NULL
g_value_init (&v, GST_TYPE_BUFFER);
gst_value_set_buffer (&v, rtpmp4gpay->config);
config = gst_value_serialize (&v);
/* hmm, silly */
if (rtpmp4gpay->params) {
res = gst_basertppayload_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),
MP4GCAPS);
}
g_value_unset (&v);
g_free (config);
#undef MP4GCAPS
return res;
}
/* sample conversion functions */
static GstClockTime
gst_base_rtp_audio_payload_sample_bytes_to_time (GstBaseRTPAudioPayload *
payload, guint64 bytes)
{
guint64 rtptime;
/* avoid division when we can */
if (G_LIKELY (payload->sample_size != 8))
rtptime = gst_util_uint64_scale_int (bytes, 8, payload->sample_size);
else
rtptime = bytes;
return gst_util_uint64_scale_int (rtptime, GST_SECOND,
GST_BASE_RTP_PAYLOAD (payload)->clock_rate);
}
static gboolean
gst_basertppayload_event (GstPad * pad, GstEvent * event)
{
GstBaseRTPPayload *basertppayload;
GstBaseRTPPayloadClass *basertppayload_class;
gboolean res;
basertppayload = GST_BASE_RTP_PAYLOAD (gst_pad_get_parent (pad));
basertppayload_class = GST_BASE_RTP_PAYLOAD_GET_CLASS (basertppayload);
if (basertppayload_class->handle_event) {
res = basertppayload_class->handle_event (pad, event);
if (res)
goto done;
}
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_FLUSH_START:
res = gst_pad_event_default (pad, event);
break;
case GST_EVENT_FLUSH_STOP:
res = gst_pad_event_default (pad, event);
gst_segment_init (&basertppayload->segment, GST_FORMAT_UNDEFINED);
break;
case GST_EVENT_NEWSEGMENT:
{
gboolean update;
gdouble rate;
GstFormat fmt;
gint64 start, stop, position;
gst_event_parse_new_segment (event, &update, &rate, &fmt, &start, &stop,
&position);
gst_segment_set_newsegment (&basertppayload->segment, update, rate, fmt,
start, stop, position);
/* fallthrough */
}
default:
res = gst_pad_event_default (pad, event);
break;
}
done:
gst_object_unref (basertppayload);
return res;
}
static guint32
gst_base_rtp_audio_payload_frame_bytes_to_rtptime (GstBaseRTPAudioPayload *
payload, guint64 bytes)
{
guint64 framecount;
guint64 time;
framecount = bytes / payload->frame_size;
if (G_UNLIKELY (bytes % payload->frame_size))
framecount++;
time = framecount * payload->priv->frame_duration_ns;
return gst_util_uint64_scale_int (time,
GST_BASE_RTP_PAYLOAD (payload)->clock_rate, GST_SECOND);
}
static GstFlowReturn gst_rtp_sbc_pay_flush_buffers(GstRtpSBCPay *sbcpay)
{
guint available;
guint max_payload;
GstBuffer *outbuf;
guint8 *payload_data;
guint frame_count;
guint payload_length;
struct rtp_payload *payload;
if (sbcpay->frame_length == 0) {
GST_ERROR_OBJECT(sbcpay, "Frame length is 0");
return GST_FLOW_ERROR;
}
available = gst_adapter_available(sbcpay->adapter);
max_payload = gst_rtp_buffer_calc_payload_len(
GST_BASE_RTP_PAYLOAD_MTU(sbcpay) - RTP_SBC_PAYLOAD_HEADER_SIZE,
0, 0);
max_payload = MIN(max_payload, available);
frame_count = max_payload / sbcpay->frame_length;
payload_length = frame_count * sbcpay->frame_length;
if (payload_length == 0) /* Nothing to send */
return GST_FLOW_OK;
outbuf = gst_rtp_buffer_new_allocate(payload_length +
RTP_SBC_PAYLOAD_HEADER_SIZE, 0, 0);
gst_rtp_buffer_set_payload_type(outbuf,
GST_BASE_RTP_PAYLOAD_PT(sbcpay));
payload_data = gst_rtp_buffer_get_payload(outbuf);
payload = (struct rtp_payload *) payload_data;
memset(payload, 0, sizeof(struct rtp_payload));
payload->frame_count = frame_count;
gst_adapter_copy(sbcpay->adapter, payload_data +
RTP_SBC_PAYLOAD_HEADER_SIZE, 0, payload_length);
gst_adapter_flush(sbcpay->adapter, payload_length);
GST_BUFFER_TIMESTAMP(outbuf) = sbcpay->timestamp;
GST_DEBUG_OBJECT(sbcpay, "Pushing %d bytes", payload_length);
return gst_basertppayload_push(GST_BASE_RTP_PAYLOAD(sbcpay), outbuf);
}
static void
gst_rtp_siren_pay_init (GstRTPSirenPay * rtpsirenpay,
GstRTPSirenPayClass * klass)
{
GstBaseRTPPayload *basertppayload;
GstBaseRTPAudioPayload *basertpaudiopayload;
basertppayload = GST_BASE_RTP_PAYLOAD (rtpsirenpay);
basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (rtpsirenpay);
/* we don't set the payload type, it should be set by the application using
* the pt property or the default 96 will be used */
basertppayload->clock_rate = 16000;
/* tell basertpaudiopayload that this is a frame based codec */
gst_base_rtp_audio_payload_set_frame_based (basertpaudiopayload);
}
static void
gst_basertppayload_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstBaseRTPPayload *basertppayload;
GstBaseRTPPayloadPrivate *priv;
gint64 val;
basertppayload = GST_BASE_RTP_PAYLOAD (object);
priv = basertppayload->priv;
switch (prop_id) {
case PROP_MTU:
basertppayload->mtu = g_value_get_uint (value);
break;
case PROP_PT:
basertppayload->pt = g_value_get_uint (value);
break;
case PROP_SSRC:
val = g_value_get_uint (value);
basertppayload->ssrc = val;
priv->ssrc_random = FALSE;
break;
case PROP_TIMESTAMP_OFFSET:
val = g_value_get_uint (value);
basertppayload->ts_offset = val;
priv->ts_offset_random = FALSE;
break;
case PROP_SEQNUM_OFFSET:
val = g_value_get_int (value);
basertppayload->seqnum_offset = val;
priv->seqnum_offset_random = (val == -1);
GST_DEBUG_OBJECT (basertppayload, "seqnum offset 0x%04x, random %d",
basertppayload->seqnum_offset, priv->seqnum_offset_random);
break;
case PROP_MAX_PTIME:
basertppayload->max_ptime = g_value_get_int64 (value);
break;
case PROP_MIN_PTIME:
basertppayload->min_ptime = g_value_get_int64 (value);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static gboolean
gst_basertppayload_sink_setcaps (GstPad * pad, GstCaps * caps)
{
GstBaseRTPPayload *basertppayload;
GstBaseRTPPayloadClass *basertppayload_class;
gboolean ret = TRUE;
GST_DEBUG_OBJECT (pad, "setting caps %" GST_PTR_FORMAT, caps);
basertppayload = GST_BASE_RTP_PAYLOAD (gst_pad_get_parent (pad));
basertppayload_class = GST_BASE_RTP_PAYLOAD_GET_CLASS (basertppayload);
if (basertppayload_class->set_caps)
ret = basertppayload_class->set_caps (basertppayload, caps);
gst_object_unref (basertppayload);
return ret;
}
static GstFlowReturn
gst_rtp_asf_pay_parse_headers (GstRtpAsfPay * rtpasfpay)
{
gchar *maxps;
g_return_val_if_fail (rtpasfpay->headers, GST_FLOW_ERROR);
if (!gst_asf_parse_headers (rtpasfpay->headers, &rtpasfpay->asfinfo))
goto error;
GST_DEBUG_OBJECT (rtpasfpay, "Packets number: %" G_GUINT64_FORMAT,
rtpasfpay->asfinfo.packets_count);
GST_DEBUG_OBJECT (rtpasfpay, "Packets size: %" G_GUINT32_FORMAT,
rtpasfpay->asfinfo.packet_size);
GST_DEBUG_OBJECT (rtpasfpay, "Broadcast mode: %s",
rtpasfpay->asfinfo.broadcast ? "true" : "false");
/* get the config for caps */
g_free (rtpasfpay->config);
rtpasfpay->config = g_base64_encode (GST_BUFFER_DATA (rtpasfpay->headers),
GST_BUFFER_SIZE (rtpasfpay->headers));
GST_DEBUG_OBJECT (rtpasfpay, "Serialized headers to base64 string %s",
rtpasfpay->config);
g_assert (rtpasfpay->config != NULL);
GST_DEBUG_OBJECT (rtpasfpay, "Setting optional caps values: maxps=%"
G_GUINT32_FORMAT " and config=%s", rtpasfpay->asfinfo.packet_size,
rtpasfpay->config);
maxps =
g_strdup_printf ("%" G_GUINT32_FORMAT, rtpasfpay->asfinfo.packet_size);
gst_basertppayload_set_outcaps (GST_BASE_RTP_PAYLOAD (rtpasfpay), "maxps",
G_TYPE_STRING, maxps, "config", G_TYPE_STRING, rtpasfpay->config, NULL);
g_free (maxps);
return GST_FLOW_OK;
error:
{
GST_ELEMENT_ERROR (rtpasfpay, STREAM, DECODE, (NULL),
("Error parsing headers"));
return GST_FLOW_ERROR;
}
}
static gboolean gst_a2dp_sink_init_rtp_sbc_element(GstA2dpSink *self)
{
GstElement *rtppay;
/* if we already have a rtp, we don't need a new one */
if (self->rtp != NULL)
return TRUE;
rtppay = gst_a2dp_sink_init_element(self, "rtpsbcpay", "rtp",
self->capsfilter);
if (rtppay == NULL)
return FALSE;
self->rtp = GST_BASE_RTP_PAYLOAD(rtppay);
g_object_set(G_OBJECT(self->rtp), "min-frames", -1, NULL);
gst_element_set_state(rtppay, GST_STATE_PAUSED);
return TRUE;
}
static void
gst_rtp_mp4v_pay_new_caps (GstRtpMP4VPay * rtpmp4vpay)
{
gchar *profile, *config;
GValue v = { 0 };
profile = g_strdup_printf ("%d", rtpmp4vpay->profile);
g_value_init (&v, GST_TYPE_BUFFER);
gst_value_set_buffer (&v, rtpmp4vpay->config);
config = gst_value_serialize (&v);
gst_basertppayload_set_outcaps (GST_BASE_RTP_PAYLOAD (rtpmp4vpay),
"profile-level-id", G_TYPE_STRING, profile,
"config", G_TYPE_STRING, config, NULL);
g_value_unset (&v);
g_free (profile);
g_free (config);
}
static void
gst_basertppayload_finalize (GObject * object)
{
GstBaseRTPPayload *basertppayload;
basertppayload = GST_BASE_RTP_PAYLOAD (object);
g_rand_free (basertppayload->seq_rand);
basertppayload->seq_rand = NULL;
g_rand_free (basertppayload->ssrc_rand);
basertppayload->ssrc_rand = NULL;
g_rand_free (basertppayload->ts_rand);
basertppayload->ts_rand = NULL;
g_free (basertppayload->media);
basertppayload->media = NULL;
g_free (basertppayload->encoding_name);
basertppayload->encoding_name = NULL;
G_OBJECT_CLASS (parent_class)->finalize (object);
}