static gboolean
gst_dv_pay_negotiate (GstRTPDVPay * rtpdvpay, guint8 * data, gsize 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_rtp_base_payload_set_options (GST_RTP_BASE_PAYLOAD (rtpdvpay), media,
TRUE, "DV", 90000);
if (audio_bundled) {
res = gst_rtp_base_payload_set_outcaps (GST_RTP_BASE_PAYLOAD (rtpdvpay),
"encode", G_TYPE_STRING, encode,
"audio", G_TYPE_STRING, "bundled", NULL);
} else {
res = gst_rtp_base_payload_set_outcaps (GST_RTP_BASE_PAYLOAD (rtpdvpay),
"encode", G_TYPE_STRING, encode, NULL);
}
return res;
}
static void
gst_rtp_pcma_pay_init (GstRtpPcmaPay * rtppcmapay)
{
GstRTPBaseAudioPayload *rtpbaseaudiopayload;
rtpbaseaudiopayload = GST_RTP_BASE_AUDIO_PAYLOAD (rtppcmapay);
GST_RTP_BASE_PAYLOAD (rtppcmapay)->pt = GST_RTP_PAYLOAD_PCMA;
GST_RTP_BASE_PAYLOAD (rtppcmapay)->clock_rate = 8000;
/* tell rtpbaseaudiopayload that this is a sample based codec */
gst_rtp_base_audio_payload_set_sample_based (rtpbaseaudiopayload);
/* octet-per-sample is 1 for PCM */
gst_rtp_base_audio_payload_set_sample_options (rtpbaseaudiopayload, 1);
}
static GstFlowReturn
gst_rtp_base_payload_chain (GstPad * pad, GstObject * parent,
GstBuffer * buffer)
{
GstRTPBasePayload *rtpbasepayload;
GstRTPBasePayloadClass *rtpbasepayload_class;
GstFlowReturn ret;
rtpbasepayload = GST_RTP_BASE_PAYLOAD (parent);
rtpbasepayload_class = GST_RTP_BASE_PAYLOAD_GET_CLASS (rtpbasepayload);
if (!rtpbasepayload_class->handle_buffer)
goto no_function;
ret = rtpbasepayload_class->handle_buffer (rtpbasepayload, buffer);
return ret;
/* ERRORS */
no_function:
{
GST_ELEMENT_ERROR (rtpbasepayload, STREAM, NOT_IMPLEMENTED, (NULL),
("subclass did not implement handle_buffer function"));
gst_buffer_unref (buffer);
return GST_FLOW_ERROR;
}
}
static void
gst_rtp_mpa_pay_init (GstRtpMPAPay * rtpmpapay)
{
rtpmpapay->adapter = gst_adapter_new ();
GST_RTP_BASE_PAYLOAD (rtpmpapay)->pt = GST_RTP_PAYLOAD_MPA;
}
/**
* gst_rtp_base_audio_payload_push:
* @baseaudiopayload: a #GstRTPBasePayload
* @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
*/
GstFlowReturn
gst_rtp_base_audio_payload_push (GstRTPBaseAudioPayload * baseaudiopayload,
const guint8 * data, guint payload_len, GstClockTime timestamp)
{
GstRTPBasePayload *basepayload;
GstBuffer *outbuf;
guint8 *payload;
GstFlowReturn ret;
GstRTPBuffer rtp = { NULL };
basepayload = GST_RTP_BASE_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_base_payload_allocate_output_buffer (basepayload,
payload_len, 0, 0);
/* copy payload */
gst_rtp_buffer_map (outbuf, GST_MAP_WRITE, &rtp);
payload = gst_rtp_buffer_get_payload (&rtp);
memcpy (payload, data, payload_len);
gst_rtp_buffer_unmap (&rtp);
/* set metadata */
gst_rtp_base_audio_payload_set_meta (baseaudiopayload, outbuf, payload_len,
timestamp);
ret = gst_rtp_base_payload_push (basepayload, 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;
GstRTPBuffer rtp = { NULL, };
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;
gst_rtp_buffer_map (outbuf, GST_MAP_WRITE, &rtp);
/* point to the payload for size headers and data */
spayload = gst_rtp_buffer_get_payload (&rtp);
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_get_size (buf);
while (size > 0xff) {
*spayload++ = 0xff;
size -= 0xff;
}
*spayload++ = size;
/* copy payload */
size = gst_buffer_get_size (buf);
gst_buffer_extract (buf, 0, payload, size);
payload += size;
gst_buffer_unref (buf);
}
gst_rtp_buffer_unmap (&rtp);
/* we consumed it all */
rtpceltpay->bytes = 0;
rtpceltpay->sbytes = 0;
rtpceltpay->qduration = 0;
ret = gst_rtp_base_payload_push (GST_RTP_BASE_PAYLOAD (rtpceltpay), outbuf);
return ret;
}
static void
gst_rtp_mp2t_pay_init (GstRTPMP2TPay * rtpmp2tpay)
{
GST_RTP_BASE_PAYLOAD (rtpmp2tpay)->clock_rate = 90000;
GST_RTP_BASE_PAYLOAD_PT (rtpmp2tpay) = GST_RTP_PAYLOAD_MP2T;
rtpmp2tpay->adapter = gst_adapter_new ();
}
static void
gst_rtp_base_payload_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstRTPBasePayload *rtpbasepayload;
GstRTPBasePayloadPrivate *priv;
rtpbasepayload = GST_RTP_BASE_PAYLOAD (object);
priv = rtpbasepayload->priv;
switch (prop_id) {
case PROP_MTU:
g_value_set_uint (value, rtpbasepayload->mtu);
break;
case PROP_PT:
g_value_set_uint (value, rtpbasepayload->pt);
break;
case PROP_SSRC:
if (priv->ssrc_random)
g_value_set_uint (value, -1);
else
g_value_set_uint (value, rtpbasepayload->ssrc);
break;
case PROP_TIMESTAMP_OFFSET:
if (priv->ts_offset_random)
g_value_set_uint (value, -1);
else
g_value_set_uint (value, (guint32) rtpbasepayload->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) rtpbasepayload->seqnum_offset);
break;
case PROP_MAX_PTIME:
g_value_set_int64 (value, rtpbasepayload->max_ptime);
break;
case PROP_MIN_PTIME:
g_value_set_int64 (value, rtpbasepayload->min_ptime);
break;
case PROP_TIMESTAMP:
g_value_set_uint (value, rtpbasepayload->timestamp);
break;
case PROP_SEQNUM:
g_value_set_uint (value, rtpbasepayload->seqnum);
break;
case PROP_PERFECT_RTPTIME:
g_value_set_boolean (value, priv->perfect_rtptime);
break;
case PROP_PTIME_MULTIPLE:
g_value_set_int64 (value, rtpbasepayload->ptime_multiple);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_rtp_g723_pay_init (GstRTPG723Pay * pay)
{
GstRTPBasePayload *payload = GST_RTP_BASE_PAYLOAD (pay);
pay->adapter = gst_adapter_new ();
payload->pt = GST_RTP_PAYLOAD_G723;
}
static GstFlowReturn
gst_rtp_sbc_pay_flush_buffers (GstRtpSBCPay * sbcpay)
{
GstRTPBuffer rtp = GST_RTP_BUFFER_INIT;
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_RTP_BASE_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);
/* get payload */
gst_rtp_buffer_map (outbuf, GST_MAP_WRITE, &rtp);
gst_rtp_buffer_set_payload_type (&rtp, GST_RTP_BASE_PAYLOAD_PT (sbcpay));
/* write header and copy data into payload */
payload_data = gst_rtp_buffer_get_payload (&rtp);
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_rtp_buffer_unmap (&rtp);
gst_adapter_flush (sbcpay->adapter, payload_length);
/* FIXME: what about duration? */
GST_BUFFER_PTS (outbuf) = sbcpay->timestamp;
GST_DEBUG_OBJECT (sbcpay, "Pushing %d bytes", payload_length);
return gst_rtp_base_payload_push (GST_RTP_BASE_PAYLOAD (sbcpay), outbuf);
}
static GstFlowReturn
gst_rtp_base_audio_payload_push_buffer (GstRTPBaseAudioPayload *
baseaudiopayload, GstBuffer * buffer, GstClockTime timestamp)
{
GstRTPBasePayload *basepayload;
GstRTPBaseAudioPayloadPrivate *priv;
GstBuffer *outbuf;
guint payload_len;
GstFlowReturn ret;
priv = baseaudiopayload->priv;
basepayload = GST_RTP_BASE_PAYLOAD (baseaudiopayload);
payload_len = gst_buffer_get_size (buffer);
GST_DEBUG_OBJECT (baseaudiopayload, "Pushing %d bytes ts %" GST_TIME_FORMAT,
payload_len, GST_TIME_ARGS (timestamp));
/* create just the RTP header buffer */
outbuf = gst_rtp_base_payload_allocate_output_buffer (basepayload, 0, 0, 0);
/* set metadata */
gst_rtp_base_audio_payload_set_meta (baseaudiopayload, outbuf, payload_len,
timestamp);
if (priv->buffer_list) {
GstBufferList *list;
guint i, len;
list = gst_buffer_list_new ();
len = gst_buffer_list_length (list);
for (i = 0; i < len; i++) {
/* FIXME */
g_warning ("bufferlist not implemented");
gst_buffer_list_add (list, outbuf);
gst_buffer_list_add (list, buffer);
}
GST_DEBUG_OBJECT (baseaudiopayload, "Pushing list %p", list);
ret = gst_rtp_base_payload_push_list (basepayload, list);
} else {
CopyMetaData data;
/* copy payload */
data.pay = baseaudiopayload;
data.outbuf = outbuf;
gst_buffer_foreach_meta (buffer, foreach_metadata, &data);
outbuf = gst_buffer_append (outbuf, buffer);
GST_DEBUG_OBJECT (baseaudiopayload, "Pushing buffer %p", outbuf);
ret = gst_rtp_base_payload_push (basepayload, outbuf);
}
return ret;
}
static void
gst_rtp_g723_pay_init (GstRTPG723Pay * pay)
{
GstRTPBasePayload *payload = GST_RTP_BASE_PAYLOAD (pay);
pay->adapter = gst_adapter_new ();
payload->pt = GST_RTP_PAYLOAD_G723;
gst_rtp_base_payload_set_options (payload, "audio", FALSE, "G723", 8000);
}
static GstStateChangeReturn
gst_rtp_base_payload_change_state (GstElement * element,
GstStateChange transition)
{
GstRTPBasePayload *rtpbasepayload;
GstRTPBasePayloadPrivate *priv;
GstStateChangeReturn ret;
rtpbasepayload = GST_RTP_BASE_PAYLOAD (element);
priv = rtpbasepayload->priv;
switch (transition) {
case GST_STATE_CHANGE_NULL_TO_READY:
break;
case GST_STATE_CHANGE_READY_TO_PAUSED:
gst_segment_init (&rtpbasepayload->segment, GST_FORMAT_UNDEFINED);
if (priv->seqnum_offset_random)
rtpbasepayload->seqnum_base = g_random_int_range (0, G_MAXUINT16);
else
rtpbasepayload->seqnum_base = rtpbasepayload->seqnum_offset;
priv->next_seqnum = rtpbasepayload->seqnum_base;
rtpbasepayload->seqnum = rtpbasepayload->seqnum_base;
if (priv->ssrc_random)
rtpbasepayload->current_ssrc = g_random_int ();
else
rtpbasepayload->current_ssrc = rtpbasepayload->ssrc;
if (priv->ts_offset_random)
rtpbasepayload->ts_base = g_random_int ();
else
rtpbasepayload->ts_base = rtpbasepayload->ts_offset;
rtpbasepayload->timestamp = rtpbasepayload->ts_base;
g_atomic_int_set (&rtpbasepayload->priv->notified_first_timestamp, 1);
priv->base_offset = GST_BUFFER_OFFSET_NONE;
break;
default:
break;
}
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
switch (transition) {
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
g_atomic_int_set (&rtpbasepayload->priv->notified_first_timestamp, 1);
break;
case GST_STATE_CHANGE_READY_TO_NULL:
break;
default:
break;
}
return ret;
}
static void
gst_rtp_base_payload_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstRTPBasePayload *rtpbasepayload;
GstRTPBasePayloadPrivate *priv;
gint64 val;
rtpbasepayload = GST_RTP_BASE_PAYLOAD (object);
priv = rtpbasepayload->priv;
switch (prop_id) {
case PROP_MTU:
rtpbasepayload->mtu = g_value_get_uint (value);
break;
case PROP_PT:
rtpbasepayload->pt = g_value_get_uint (value);
break;
case PROP_SSRC:
val = g_value_get_uint (value);
rtpbasepayload->ssrc = val;
priv->ssrc_random = FALSE;
break;
case PROP_TIMESTAMP_OFFSET:
val = g_value_get_uint (value);
rtpbasepayload->ts_offset = val;
priv->ts_offset_random = FALSE;
break;
case PROP_SEQNUM_OFFSET:
val = g_value_get_int (value);
rtpbasepayload->seqnum_offset = val;
priv->seqnum_offset_random = (val == -1);
GST_DEBUG_OBJECT (rtpbasepayload, "seqnum offset 0x%04x, random %d",
rtpbasepayload->seqnum_offset, priv->seqnum_offset_random);
break;
case PROP_MAX_PTIME:
rtpbasepayload->priv->prop_max_ptime = g_value_get_int64 (value);
update_max_ptime (rtpbasepayload);
break;
case PROP_MIN_PTIME:
rtpbasepayload->min_ptime = g_value_get_int64 (value);
break;
case PROP_PERFECT_RTPTIME:
priv->perfect_rtptime = g_value_get_boolean (value);
break;
case PROP_PTIME_MULTIPLE:
rtpbasepayload->ptime_multiple = g_value_get_int64 (value);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_rtp_g722_pay_init (GstRtpG722Pay * rtpg722pay)
{
GstRTPBaseAudioPayload *rtpbaseaudiopayload;
rtpbaseaudiopayload = GST_RTP_BASE_AUDIO_PAYLOAD (rtpg722pay);
GST_RTP_BASE_PAYLOAD (rtpg722pay)->pt = GST_RTP_PAYLOAD_G722;
/* tell rtpbaseaudiopayload that this is a sample based codec */
gst_rtp_base_audio_payload_set_sample_based (rtpbaseaudiopayload);
}
static GstFlowReturn
gst_rtp_mp2t_pay_flush (GstRTPMP2TPay * rtpmp2tpay)
{
guint avail, mtu;
GstFlowReturn ret = GST_FLOW_OK;
GstBuffer *outbuf;
avail = gst_adapter_available (rtpmp2tpay->adapter);
mtu = GST_RTP_BASE_PAYLOAD_MTU (rtpmp2tpay);
while (avail > 0 && (ret == GST_FLOW_OK)) {
guint towrite;
guint payload_len;
guint packet_len;
GstBuffer *paybuf;
/* this will be the total length 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, mtu);
/* this is the payload length */
payload_len = gst_rtp_buffer_calc_payload_len (towrite, 0, 0);
payload_len -= payload_len % 188;
/* need whole packets */
if (!payload_len)
break;
/* create buffer to hold the payload */
outbuf = gst_rtp_buffer_new_allocate (0, 0, 0);
/* get payload */
paybuf = gst_adapter_take_buffer_fast (rtpmp2tpay->adapter, payload_len);
gst_rtp_copy_meta (GST_ELEMENT_CAST (rtpmp2tpay), outbuf, paybuf, 0);
outbuf = gst_buffer_append (outbuf, paybuf);
avail -= payload_len;
GST_BUFFER_PTS (outbuf) = rtpmp2tpay->first_ts;
GST_BUFFER_DURATION (outbuf) = rtpmp2tpay->duration;
GST_DEBUG_OBJECT (rtpmp2tpay, "pushing buffer of size %u",
(guint) gst_buffer_get_size (outbuf));
ret = gst_rtp_base_payload_push (GST_RTP_BASE_PAYLOAD (rtpmp2tpay), outbuf);
}
return ret;
}
static void
gst_rtp_base_payload_finalize (GObject * object)
{
GstRTPBasePayload *rtpbasepayload;
rtpbasepayload = GST_RTP_BASE_PAYLOAD (object);
g_free (rtpbasepayload->media);
rtpbasepayload->media = NULL;
g_free (rtpbasepayload->encoding_name);
rtpbasepayload->encoding_name = NULL;
G_OBJECT_CLASS (parent_class)->finalize (object);
}
/* sample conversion functions */
static GstClockTime
gst_rtp_base_audio_payload_sample_bytes_to_time (GstRTPBaseAudioPayload *
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_RTP_BASE_PAYLOAD (payload)->clock_rate);
}
static guint64
gst_rtp_base_audio_payload_sample_time_to_bytes (GstRTPBaseAudioPayload *
payload, guint64 time)
{
guint64 samples;
samples = gst_util_uint64_scale_int (time,
GST_RTP_BASE_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_rtp_base_payload_set_outcaps (GST_RTP_BASE_PAYLOAD (rtpmp4gpay),
"encoding-params", G_TYPE_STRING, rtpmp4gpay->params, MP4GCAPS);
} else {
res = gst_rtp_base_payload_set_outcaps (GST_RTP_BASE_PAYLOAD (rtpmp4gpay),
MP4GCAPS);
}
g_value_unset (&v);
g_free (config);
#undef MP4GCAPS
return res;
}
static gboolean
gst_rtp_base_payload_query (GstPad * pad, GstObject * parent, GstQuery * query)
{
GstRTPBasePayload *rtpbasepayload;
GstRTPBasePayloadClass *rtpbasepayload_class;
gboolean res = FALSE;
rtpbasepayload = GST_RTP_BASE_PAYLOAD (parent);
rtpbasepayload_class = GST_RTP_BASE_PAYLOAD_GET_CLASS (rtpbasepayload);
if (rtpbasepayload_class->query)
res = rtpbasepayload_class->query (rtpbasepayload, pad, query);
return res;
}
static void
gst_rtp_siren_pay_init (GstRTPSirenPay * rtpsirenpay)
{
GstRTPBasePayload *rtpbasepayload;
GstRTPBaseAudioPayload *rtpbaseaudiopayload;
rtpbasepayload = GST_RTP_BASE_PAYLOAD (rtpsirenpay);
rtpbaseaudiopayload = GST_RTP_BASE_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 */
rtpbasepayload->clock_rate = 16000;
/* tell rtpbaseaudiopayload that this is a frame based codec */
gst_rtp_base_audio_payload_set_frame_based (rtpbaseaudiopayload);
}
static guint32
gst_rtp_base_audio_payload_frame_bytes_to_rtptime (GstRTPBaseAudioPayload *
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_RTP_BASE_PAYLOAD (payload)->clock_rate, GST_SECOND);
}
static GstFlowReturn
gst_rtp_asf_pay_parse_headers (GstRtpAsfPay * rtpasfpay)
{
gchar *maxps;
GstMapInfo map;
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);
gst_buffer_map (rtpasfpay->headers, &map, GST_MAP_READ);
rtpasfpay->config = g_base64_encode (map.data, map.size);
gst_buffer_unmap (rtpasfpay->headers, &map);
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_rtp_base_payload_set_outcaps (GST_RTP_BASE_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_rtp_base_payload_sink_event (GstPad * pad, GstObject * parent,
GstEvent * event)
{
GstRTPBasePayload *rtpbasepayload;
GstRTPBasePayloadClass *rtpbasepayload_class;
gboolean res = FALSE;
rtpbasepayload = GST_RTP_BASE_PAYLOAD (parent);
rtpbasepayload_class = GST_RTP_BASE_PAYLOAD_GET_CLASS (rtpbasepayload);
if (rtpbasepayload_class->sink_event)
res = rtpbasepayload_class->sink_event (rtpbasepayload, event);
else
gst_event_unref (event);
return res;
}
static GstFlowReturn
gst_rtp_g723_pay_flush (GstRTPG723Pay * pay)
{
GstBuffer *outbuf;
GstFlowReturn ret;
guint8 *payload;
guint avail;
GstRTPBuffer rtp = { NULL };
avail = gst_adapter_available (pay->adapter);
outbuf = gst_rtp_buffer_new_allocate (avail, 0, 0);
gst_rtp_buffer_map (outbuf, GST_MAP_WRITE, &rtp);
payload = gst_rtp_buffer_get_payload (&rtp);
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 (&rtp, TRUE);
pay->discont = FALSE;
}
gst_rtp_buffer_unmap (&rtp);
ret = gst_rtp_base_payload_push (GST_RTP_BASE_PAYLOAD (pay), outbuf);
return ret;
}
static GstFlowReturn
gst_rtp_g723_pay_flush (GstRTPG723Pay * pay)
{
GstBuffer *outbuf, *payload_buf;
GstFlowReturn ret;
guint avail;
GstRTPBuffer rtp = { NULL };
avail = gst_adapter_available (pay->adapter);
outbuf = gst_rtp_buffer_new_allocate (0, 0, 0);
gst_rtp_buffer_map (outbuf, GST_MAP_WRITE, &rtp);
GST_BUFFER_PTS (outbuf) = pay->timestamp;
GST_BUFFER_DURATION (outbuf) = pay->duration;
/* copy G723 data as payload */
payload_buf = gst_adapter_take_buffer_fast (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 (&rtp, TRUE);
pay->discont = FALSE;
}
gst_rtp_buffer_unmap (&rtp);
gst_rtp_copy_meta (GST_ELEMENT_CAST (pay), outbuf, payload_buf,
g_quark_from_static_string (GST_META_TAG_AUDIO_STR));
outbuf = gst_buffer_append (outbuf, payload_buf);
ret = gst_rtp_base_payload_push (GST_RTP_BASE_PAYLOAD (pay), outbuf);
return ret;
}
static GstFlowReturn
gst_rtp_asf_pay_handle_packet (GstRtpAsfPay * rtpasfpay, GstBuffer * buffer)
{
GstRTPBasePayload *rtppay;
GstAsfPacketInfo *packetinfo;
guint8 flags;
guint8 *data;
guint32 packet_util_size;
guint32 packet_offset;
guint32 size_left;
GstFlowReturn ret = GST_FLOW_OK;
rtppay = GST_RTP_BASE_PAYLOAD (rtpasfpay);
packetinfo = &rtpasfpay->packetinfo;
if (!gst_asf_parse_packet (buffer, packetinfo, TRUE,
rtpasfpay->asfinfo.packet_size)) {
GST_ERROR_OBJECT (rtpasfpay, "Error while parsing asf packet");
gst_buffer_unref (buffer);
return GST_FLOW_ERROR;
}
if (packetinfo->packet_size == 0)
packetinfo->packet_size = rtpasfpay->asfinfo.packet_size;
GST_LOG_OBJECT (rtpasfpay, "Packet size: %" G_GUINT32_FORMAT
", padding: %" G_GUINT32_FORMAT, packetinfo->packet_size,
packetinfo->padding);
/* update padding field to 0 */
if (packetinfo->padding > 0) {
GstAsfPacketInfo info;
/* find padding field offset */
guint offset = packetinfo->err_cor_len + 2 +
gst_asf_get_var_size_field_len (packetinfo->packet_field_type) +
gst_asf_get_var_size_field_len (packetinfo->seq_field_type);
buffer = gst_buffer_make_writable (buffer);
switch (packetinfo->padd_field_type) {
case ASF_FIELD_TYPE_DWORD:
gst_buffer_memset (buffer, offset, 0, 4);
break;
case ASF_FIELD_TYPE_WORD:
gst_buffer_memset (buffer, offset, 0, 2);
break;
case ASF_FIELD_TYPE_BYTE:
gst_buffer_memset (buffer, offset, 0, 1);
break;
case ASF_FIELD_TYPE_NONE:
default:
break;
}
gst_asf_parse_packet (buffer, &info, FALSE, 0);
}
if (packetinfo->padding != 0)
packet_util_size = rtpasfpay->asfinfo.packet_size - packetinfo->padding;
else
packet_util_size = packetinfo->packet_size;
packet_offset = 0;
while (packet_util_size > 0) {
/* Even if we don't fill completely an output buffer we
* push it when we add an fragment. Because it seems that
* it is not possible to determine where a asf packet
* fragment ends inside a rtp packet payload.
* This flag tells us to push the packet.
*/
gboolean force_push = FALSE;
GstRTPBuffer rtp = GST_RTP_BUFFER_INIT;
/* we have no output buffer pending, create one */
if (rtpasfpay->current == NULL) {
GST_LOG_OBJECT (rtpasfpay, "Creating new output buffer");
rtpasfpay->current =
gst_rtp_buffer_new_allocate_len (GST_RTP_BASE_PAYLOAD_MTU (rtpasfpay),
0, 0);
rtpasfpay->cur_off = 0;
rtpasfpay->has_ts = FALSE;
rtpasfpay->marker = FALSE;
}
gst_rtp_buffer_map (rtpasfpay->current, GST_MAP_READWRITE, &rtp);
data = gst_rtp_buffer_get_payload (&rtp);
data += rtpasfpay->cur_off;
size_left = gst_rtp_buffer_get_payload_len (&rtp) - rtpasfpay->cur_off;
GST_DEBUG_OBJECT (rtpasfpay, "Input buffer bytes consumed: %"
G_GUINT32_FORMAT "/%" G_GSIZE_FORMAT, packet_offset,
gst_buffer_get_size (buffer));
GST_DEBUG_OBJECT (rtpasfpay, "Output rtpbuffer status");
GST_DEBUG_OBJECT (rtpasfpay, "Current offset: %" G_GUINT32_FORMAT,
rtpasfpay->cur_off);
GST_DEBUG_OBJECT (rtpasfpay, "Size left: %" G_GUINT32_FORMAT, size_left);
GST_DEBUG_OBJECT (rtpasfpay, "Has ts: %s",
rtpasfpay->has_ts ? "yes" : "no");
if (rtpasfpay->has_ts) {
GST_DEBUG_OBJECT (rtpasfpay, "Ts: %" G_GUINT32_FORMAT, rtpasfpay->ts);
}
flags = 0;
if (packetinfo->has_keyframe) {
flags = flags | 0x80;
}
flags = flags | 0x20; /* Relative timestamp is present */
if (!rtpasfpay->has_ts) {
/* this is the first asf packet, its send time is the
* rtp packet timestamp */
rtpasfpay->has_ts = TRUE;
rtpasfpay->ts = packetinfo->send_time;
}
if (size_left >= packet_util_size + 8) {
/* enough space for the rest of the packet */
if (packet_offset == 0) {
flags = flags | 0x40;
GST_WRITE_UINT24_BE (data + 1, packet_util_size);
} else {
GST_WRITE_UINT24_BE (data + 1, packet_offset);
force_push = TRUE;
}
data[0] = flags;
GST_WRITE_UINT32_BE (data + 4,
(gint32) (packetinfo->send_time) - (gint32) rtpasfpay->ts);
gst_buffer_extract (buffer, packet_offset, data + 8, packet_util_size);
/* updating status variables */
rtpasfpay->cur_off += 8 + packet_util_size;
size_left -= packet_util_size + 8;
packet_offset += packet_util_size;
packet_util_size = 0;
rtpasfpay->marker = TRUE;
} else {
/* fragment packet */
data[0] = flags;
GST_WRITE_UINT24_BE (data + 1, packet_offset);
GST_WRITE_UINT32_BE (data + 4,
(gint32) (packetinfo->send_time) - (gint32) rtpasfpay->ts);
gst_buffer_extract (buffer, packet_offset, data + 8, size_left - 8);
/* updating status variables */
rtpasfpay->cur_off += size_left;
packet_offset += size_left - 8;
packet_util_size -= size_left - 8;
size_left = 0;
force_push = TRUE;
}
/* there is not enough room for any more buffers */
if (force_push || size_left <= 8) {
gst_rtp_buffer_set_ssrc (&rtp, rtppay->current_ssrc);
gst_rtp_buffer_set_marker (&rtp, rtpasfpay->marker);
gst_rtp_buffer_set_payload_type (&rtp, GST_RTP_BASE_PAYLOAD_PT (rtppay));
gst_rtp_buffer_set_seq (&rtp, rtppay->seqnum + 1);
gst_rtp_buffer_set_timestamp (&rtp, packetinfo->send_time);
gst_rtp_buffer_unmap (&rtp);
/* trim remaining bytes not used */
if (size_left != 0) {
gst_buffer_set_size (rtpasfpay->current,
gst_buffer_get_size (rtpasfpay->current) - size_left);
}
GST_BUFFER_TIMESTAMP (rtpasfpay->current) = GST_BUFFER_TIMESTAMP (buffer);
rtppay->seqnum++;
rtppay->timestamp = packetinfo->send_time;
GST_DEBUG_OBJECT (rtpasfpay, "Pushing rtp buffer");
ret = gst_rtp_base_payload_push (rtppay, rtpasfpay->current);
rtpasfpay->current = NULL;
if (ret != GST_FLOW_OK) {
gst_buffer_unref (buffer);
return ret;
}
}
}
gst_buffer_unref (buffer);
return ret;
}
static GstFlowReturn
gst_rtp_mpa_pay_flush (GstRtpMPAPay * rtpmpapay)
{
guint avail;
GstBuffer *outbuf;
GstFlowReturn ret;
guint16 frag_offset;
GstBufferList *list;
/* 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 MPA data
* over multiple packets. The frag_offset in each packet header
* needs to be updated with the position in the MPA frame. */
avail = gst_adapter_available (rtpmpapay->adapter);
ret = GST_FLOW_OK;
list =
gst_buffer_list_new_sized (avail / (GST_RTP_BASE_PAYLOAD_MTU (rtpmpapay) -
RTP_HEADER_LEN) + 1);
frag_offset = 0;
while (avail > 0) {
guint towrite;
guint8 *payload;
guint payload_len;
guint packet_len;
GstRTPBuffer rtp = { NULL };
GstBuffer *paybuf;
/* this will be the total length of the packet */
packet_len = gst_rtp_buffer_calc_packet_len (4 + avail, 0, 0);
/* fill one MTU or all available bytes */
towrite = MIN (packet_len, GST_RTP_BASE_PAYLOAD_MTU (rtpmpapay));
/* 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 (4, 0, 0);
gst_rtp_buffer_map (outbuf, GST_MAP_WRITE, &rtp);
payload_len -= 4;
gst_rtp_buffer_set_payload_type (&rtp, GST_RTP_PAYLOAD_MPA);
/*
* 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 | Frag_offset |
* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
*/
payload = gst_rtp_buffer_get_payload (&rtp);
payload[0] = 0;
payload[1] = 0;
payload[2] = frag_offset >> 8;
payload[3] = frag_offset & 0xff;
avail -= payload_len;
frag_offset += payload_len;
if (avail == 0)
gst_rtp_buffer_set_marker (&rtp, TRUE);
gst_rtp_buffer_unmap (&rtp);
paybuf = gst_adapter_take_buffer_fast (rtpmpapay->adapter, payload_len);
gst_rtp_copy_meta (GST_ELEMENT_CAST (rtpmpapay), outbuf, paybuf,
g_quark_from_static_string (GST_META_TAG_AUDIO_STR));
outbuf = gst_buffer_append (outbuf, paybuf);
GST_BUFFER_PTS (outbuf) = rtpmpapay->first_ts;
GST_BUFFER_DURATION (outbuf) = rtpmpapay->duration;
gst_buffer_list_add (list, outbuf);
}
ret = gst_rtp_base_payload_push_list (GST_RTP_BASE_PAYLOAD (rtpmpapay), list);
return ret;
}
static gboolean
gst_rtp_celt_pay_parse_ident (GstRtpCELTPay * rtpceltpay,
const guint8 * data, guint size)
{
guint32 version, header_size, rate, nb_channels, frame_size, overlap;
guint32 bytes_per_packet;
GstRTPBasePayload *payload;
gchar *cstr, *fsstr;
gboolean res;
/* we need the header string (8), the version string (20), the version
* and the header length. */
if (size < 36)
goto too_small;
if (!g_str_has_prefix ((const gchar *) data, "CELT "))
goto wrong_header;
/* skip header and version string */
data += 28;
version = GST_READ_UINT32_LE (data);
GST_DEBUG_OBJECT (rtpceltpay, "version %08x", version);
#if 0
if (version != 1)
goto wrong_version;
#endif
data += 4;
/* ensure sizes */
header_size = GST_READ_UINT32_LE (data);
if (header_size < 56)
goto header_too_small;
if (size < header_size)
goto payload_too_small;
data += 4;
rate = GST_READ_UINT32_LE (data);
data += 4;
nb_channels = GST_READ_UINT32_LE (data);
data += 4;
frame_size = GST_READ_UINT32_LE (data);
data += 4;
overlap = GST_READ_UINT32_LE (data);
data += 4;
bytes_per_packet = GST_READ_UINT32_LE (data);
GST_DEBUG_OBJECT (rtpceltpay, "rate %d, nb_channels %d, frame_size %d",
rate, nb_channels, frame_size);
GST_DEBUG_OBJECT (rtpceltpay, "overlap %d, bytes_per_packet %d",
overlap, bytes_per_packet);
payload = GST_RTP_BASE_PAYLOAD (rtpceltpay);
gst_rtp_base_payload_set_options (payload, "audio", FALSE, "CELT", rate);
cstr = g_strdup_printf ("%d", nb_channels);
fsstr = g_strdup_printf ("%d", frame_size);
res = gst_rtp_base_payload_set_outcaps (payload, "encoding-params",
G_TYPE_STRING, cstr, "frame-size", G_TYPE_STRING, fsstr, NULL);
g_free (cstr);
g_free (fsstr);
return res;
/* ERRORS */
too_small:
{
GST_DEBUG_OBJECT (rtpceltpay,
"ident packet too small, need at least 32 bytes");
return FALSE;
}
wrong_header:
{
GST_DEBUG_OBJECT (rtpceltpay,
"ident packet does not start with \"CELT \"");
return FALSE;
}
#if 0
wrong_version:
{
GST_DEBUG_OBJECT (rtpceltpay, "can only handle version 1, have version %d",
version);
return FALSE;
}
#endif
header_too_small:
{
GST_DEBUG_OBJECT (rtpceltpay,
"header size too small, need at least 80 bytes, " "got only %d",
header_size);
return FALSE;
}
payload_too_small:
{
GST_DEBUG_OBJECT (rtpceltpay,
"payload too small, need at least %d bytes, got only %d", header_size,
size);
return FALSE;
}
}
