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webrtc: move some functions to the appropriate files

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This commit is contained in:
Matthew Waters 2018-11-26 16:07:57 +11:00
parent d2e87e6a31
commit 5ecca0bb22
7 changed files with 230 additions and 201 deletions
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220
ext/webrtc/gstwebrtcbin.c
View file

@ -216,41 +216,6 @@ gst_webrtc_bin_pad_class_init (GstWebRTCBinPadClass * klass)
gobject_class->finalize = gst_webrtc_bin_pad_finalize;
}
static GstCaps *
_transport_stream_get_caps_for_pt (TransportStream * stream, guint pt)
{
guint i, len;
len = stream->ptmap->len;
for (i = 0; i < len; i++) {
PtMapItem *item = &g_array_index (stream->ptmap, PtMapItem, i);
if (item->pt == pt)
return item->caps;
}
return NULL;
}
static gint
_transport_stream_get_pt (TransportStream * stream, const gchar * encoding_name)
{
guint i;
gint ret = 0;
for (i = 0; i < stream->ptmap->len; i++) {
PtMapItem *item = &g_array_index (stream->ptmap, PtMapItem, i);
if (!gst_caps_is_empty (item->caps)) {
GstStructure *s = gst_caps_get_structure (item->caps, 0);
if (!g_strcmp0 (gst_structure_get_string (s, "encoding-name"),
encoding_name)) {
ret = item->pt;
break;
}
}
}
return ret;
}
static gboolean
gst_webrtcbin_sink_event (GstPad * pad, GstObject * parent, GstEvent * event)
{
@ -356,30 +321,6 @@ enum
static guint gst_webrtc_bin_signals[LAST_SIGNAL] = { 0 };
static GstWebRTCDTLSTransport *
_transceiver_get_transport (GstWebRTCRTPTransceiver * trans)
{
if (trans->sender) {
return trans->sender->transport;
} else if (trans->receiver) {
return trans->receiver->transport;
}
return NULL;
}
static GstWebRTCDTLSTransport *
_transceiver_get_rtcp_transport (GstWebRTCRTPTransceiver * trans)
{
if (trans->sender) {
return trans->sender->rtcp_transport;
} else if (trans->receiver) {
return trans->receiver->rtcp_transport;
}
return NULL;
}
typedef struct
{
guint session_id;
@ -827,7 +768,7 @@ _collate_ice_connection_states (GstWebRTCBin * webrtc)
g_object_get (stream, "rtcp-mux", &rtcp_mux, NULL);
transport = _transceiver_get_transport (rtp_trans)->transport;
transport = webrtc_transceiver_get_dtls_transport (rtp_trans)->transport;
/* get transport state */
g_object_get (transport, "state", &ice_state, NULL);
@ -835,7 +776,8 @@ _collate_ice_connection_states (GstWebRTCBin * webrtc)
if (ice_state != STATE (CLOSED))
all_closed = FALSE;
rtcp_transport = _transceiver_get_rtcp_transport (rtp_trans)->transport;
rtcp_transport =
webrtc_transceiver_get_rtcp_dtls_transport (rtp_trans)->transport;
if (!rtcp_mux && rtcp_transport && transport != rtcp_transport) {
g_object_get (rtcp_transport, "state", &ice_state, NULL);
@ -921,7 +863,7 @@ _collate_ice_gathering_states (GstWebRTCBin * webrtc)
g_object_get (stream, "rtcp-mux", &rtcp_mux, NULL);
transport = _transceiver_get_transport (rtp_trans)->transport;
transport = webrtc_transceiver_get_dtls_transport (rtp_trans)->transport;
/* get gathering state */
g_object_get (transport, "gathering-state", &ice_state, NULL);
@ -929,7 +871,8 @@ _collate_ice_gathering_states (GstWebRTCBin * webrtc)
if (ice_state != STATE (COMPLETE))
all_completed = FALSE;
rtcp_transport = _transceiver_get_rtcp_transport (rtp_trans)->transport;
rtcp_transport =
webrtc_transceiver_get_rtcp_dtls_transport (rtp_trans)->transport;
if (!rtcp_mux && rtcp_transport && rtcp_transport != transport) {
g_object_get (rtcp_transport, "gathering-state", &ice_state, NULL);
@ -988,7 +931,7 @@ _collate_peer_connection_states (GstWebRTCBin * webrtc)
continue;
g_object_get (stream, "rtcp-mux", &rtcp_mux, NULL);
transport = _transceiver_get_transport (rtp_trans);
transport = webrtc_transceiver_get_dtls_transport (rtp_trans);
/* get transport state */
g_object_get (transport, "state", &dtls_state, NULL);
@ -996,7 +939,7 @@ _collate_peer_connection_states (GstWebRTCBin * webrtc)
g_object_get (transport->transport, "state", &ice_state, NULL);
any_ice_state |= (1 << ice_state);
rtcp_transport = _transceiver_get_rtcp_transport (rtp_trans);
rtcp_transport = webrtc_transceiver_get_rtcp_dtls_transport (rtp_trans);
if (!rtcp_mux && rtcp_transport && rtcp_transport != transport) {
g_object_get (rtcp_transport, "state", &dtls_state, NULL);
@ -1516,32 +1459,6 @@ _create_transport_channel (GstWebRTCBin * webrtc, guint session_id)
return ret;
}
static gboolean
_message_media_is_datachannel (const GstSDPMessage * msg, guint media_id)
{
const GstSDPMedia *media;
if (!msg)
return FALSE;
if (gst_sdp_message_medias_len (msg) <= media_id)
return FALSE;
media = gst_sdp_message_get_media (msg, media_id);
if (g_strcmp0 (gst_sdp_media_get_media (media), "application") != 0)
return FALSE;
if (gst_sdp_media_formats_len (media) != 1)
return FALSE;
if (g_strcmp0 (gst_sdp_media_get_format (media, 0),
"webrtc-datachannel") != 0)
return FALSE;
return TRUE;
}
static TransportStream *
_get_or_create_rtp_transport_channel (GstWebRTCBin * webrtc, guint session_id)
{
@ -2423,54 +2340,6 @@ _get_rtx_target_pt_and_ssrc_from_caps (GstCaps * answer_caps, gint * target_pt,
gst_structure_get_uint (s, "ssrc", target_ssrc);
}
static gboolean
_parse_bundle (GstWebRTCBin * webrtc, GstSDPMessage * sdp, GStrv * bundled)
{
const gchar *group;
gboolean ret = FALSE;
group = gst_sdp_message_get_attribute_val (sdp, "group");
if (group && g_str_has_prefix (group, "BUNDLE ")) {
*bundled = g_strsplit (group + strlen ("BUNDLE "), " ", 0);
if (!(*bundled)[0]) {
GST_ERROR_OBJECT (webrtc,
"Invalid format for BUNDLE group, expected at least one mid (%s)",
group);
goto done;
}
} else {
ret = TRUE;
goto done;
}
ret = TRUE;
done:
return ret;
}
static gboolean
_get_bundle_index (GstSDPMessage * sdp, GStrv bundled, guint * idx)
{
gboolean ret = FALSE;
guint i;
for (i = 0; i < gst_sdp_message_medias_len (sdp); i++) {
const GstSDPMedia *media = gst_sdp_message_get_media (sdp, i);
const gchar *mid = gst_sdp_media_get_attribute_val (media, "mid");
if (!g_strcmp0 (mid, bundled[0])) {
*idx = i;
ret = TRUE;
break;
}
}
return ret;
}
/* TODO: use the options argument */
static GstSDPMessage *
_create_answer_task (GstWebRTCBin * webrtc, const GstStructure * options)
@ -2491,7 +2360,7 @@ _create_answer_task (GstWebRTCBin * webrtc, const GstStructure * options)
return NULL;
}
if (!_parse_bundle (webrtc, pending_remote->sdp, &bundled))
if (!_parse_bundle (pending_remote->sdp, &bundled))
goto out;
if (bundled) {
@ -3520,7 +3389,7 @@ _update_transceivers_from_sdp (GstWebRTCBin * webrtc, SDPSource source,
gboolean should_connect_bundle_stream = FALSE;
TransportStream *bundle_stream = NULL;
if (!_parse_bundle (webrtc, sdp->sdp, &bundled))
if (!_parse_bundle (sdp->sdp, &bundled))
goto done;
if (bundled) {
@ -3607,57 +3476,6 @@ done:
return ret;
}
static void
_get_ice_credentials_from_sdp_media (const GstSDPMessage * sdp, guint media_idx,
gchar ** ufrag, gchar ** pwd)
{
int i;
*ufrag = NULL;
*pwd = NULL;
{
/* search in the corresponding media section */
const GstSDPMedia *media = gst_sdp_message_get_media (sdp, media_idx);
const gchar *tmp_ufrag =
gst_sdp_media_get_attribute_val (media, "ice-ufrag");
const gchar *tmp_pwd = gst_sdp_media_get_attribute_val (media, "ice-pwd");
if (tmp_ufrag && tmp_pwd) {
*ufrag = g_strdup (tmp_ufrag);
*pwd = g_strdup (tmp_pwd);
return;
}
}
/* then in the sdp message itself */
for (i = 0; i < gst_sdp_message_attributes_len (sdp); i++) {
const GstSDPAttribute *attr = gst_sdp_message_get_attribute (sdp, i);
if (g_strcmp0 (attr->key, "ice-ufrag") == 0) {
g_assert (!*ufrag);
*ufrag = g_strdup (attr->value);
} else if (g_strcmp0 (attr->key, "ice-pwd") == 0) {
g_assert (!*pwd);
*pwd = g_strdup (attr->value);
}
}
if (!*ufrag && !*pwd) {
/* Check in the medias themselves. According to JSEP, they should be
* identical FIXME: only for bundle-d streams */
for (i = 0; i < gst_sdp_message_medias_len (sdp); i++) {
const GstSDPMedia *media = gst_sdp_message_get_media (sdp, i);
const gchar *tmp_ufrag =
gst_sdp_media_get_attribute_val (media, "ice-ufrag");
const gchar *tmp_pwd = gst_sdp_media_get_attribute_val (media, "ice-pwd");
if (tmp_ufrag && tmp_pwd) {
*ufrag = g_strdup (tmp_ufrag);
*pwd = g_strdup (tmp_pwd);
break;
}
}
}
}
struct set_description
{
GstPromise *promise;
@ -3700,7 +3518,7 @@ _set_description_task (GstWebRTCBin * webrtc, struct set_description *sd)
goto out;
}
if (!_parse_bundle (webrtc, sd->sdp->sdp, &bundled))
if (!_parse_bundle (sd->sdp->sdp, &bundled))
goto out;
if (bundled) {
@ -4432,7 +4250,7 @@ on_rtpbin_request_pt_map (GstElement * rtpbin, guint session_id, guint pt,
if (!stream)
goto unknown_session;
if ((ret = _transport_stream_get_caps_for_pt (stream, pt)))
if ((ret = transport_stream_get_caps_for_pt (stream, pt)))
gst_caps_ref (ret);
GST_TRACE_OBJECT (webrtc, "Found caps %" GST_PTR_FORMAT " for pt %d in "
@ -4530,15 +4348,15 @@ on_rtpbin_request_aux_receiver (GstElement * rtpbin, guint session_id,
stream = _find_transport_for_session (webrtc, session_id);
if (stream) {
red_pt = _transport_stream_get_pt (stream, "RED");
rtx_pt = _transport_stream_get_pt (stream, "RTX");
red_pt = transport_stream_get_pt (stream, "RED");
rtx_pt = transport_stream_get_pt (stream, "RTX");
}
if (red_pt || rtx_pt)
ret = gst_bin_new (NULL);
if (rtx_pt) {
GstCaps *rtx_caps = _transport_stream_get_caps_for_pt (stream, rtx_pt);
GstCaps *rtx_caps = transport_stream_get_caps_for_pt (stream, rtx_pt);
GstElement *rtx = gst_element_factory_make ("rtprtxreceive", NULL);
GstStructure *pt_map;
const GstStructure *s = gst_caps_get_structure (rtx_caps, 0);
@ -4615,7 +4433,7 @@ on_rtpbin_request_fec_decoder (GstElement * rtpbin, guint session_id,
* example)
*/
if (stream)
pt = _transport_stream_get_pt (stream, "ULPFEC");
pt = transport_stream_get_pt (stream, "ULPFEC");
if (pt) {
GST_DEBUG_OBJECT (webrtc, "Creating ULPFEC decoder for pt %d in session %u",
@ -4648,8 +4466,8 @@ on_rtpbin_request_fec_encoder (GstElement * rtpbin, guint session_id,
(FindTransceiverFunc) transceiver_match_for_mline);
if (stream) {
ulpfec_pt = _transport_stream_get_pt (stream, "ULPFEC");
red_pt = _transport_stream_get_pt (stream, "RED");
ulpfec_pt = transport_stream_get_pt (stream, "ULPFEC");
red_pt = transport_stream_get_pt (stream, "RED");
}
if (ulpfec_pt || red_pt)
@ -4657,7 +4475,7 @@ on_rtpbin_request_fec_encoder (GstElement * rtpbin, guint session_id,
if (ulpfec_pt) {
GstElement *fecenc = gst_element_factory_make ("rtpulpfecenc", NULL);
GstCaps *caps = _transport_stream_get_caps_for_pt (stream, ulpfec_pt);
GstCaps *caps = transport_stream_get_caps_for_pt (stream, ulpfec_pt);
GST_DEBUG_OBJECT (webrtc,
"Creating ULPFEC encoder for session %d with pt %d", session_id,

35
ext/webrtc/transportstream.c
View file

@ -40,6 +40,41 @@ enum
PROP_DTLS_CLIENT,
};
GstCaps *
transport_stream_get_caps_for_pt (TransportStream * stream, guint pt)
{
guint i, len;
len = stream->ptmap->len;
for (i = 0; i < len; i++) {
PtMapItem *item = &g_array_index (stream->ptmap, PtMapItem, i);
if (item->pt == pt)
return item->caps;
}
return NULL;
}
int
transport_stream_get_pt (TransportStream * stream, const gchar * encoding_name)
{
guint i;
gint ret = 0;
for (i = 0; i < stream->ptmap->len; i++) {
PtMapItem *item = &g_array_index (stream->ptmap, PtMapItem, i);
if (!gst_caps_is_empty (item->caps)) {
GstStructure *s = gst_caps_get_structure (item->caps, 0);
if (!g_strcmp0 (gst_structure_get_string (s, "encoding-name"),
encoding_name)) {
ret = item->pt;
break;
}
}
}
return ret;
}
static void
transport_stream_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)

4
ext/webrtc/transportstream.h
View file

@ -70,6 +70,10 @@ struct _TransportStreamClass
TransportStream * transport_stream_new (GstWebRTCBin * webrtc,
guint session_id);
int transport_stream_get_pt (TransportStream * stream,
const gchar * encoding_name);
GstCaps * transport_stream_get_caps_for_pt (TransportStream * stream,
guint pt);
G_END_DECLS

124
ext/webrtc/webrtcsdp.c
View file

@ -736,3 +736,127 @@ _get_sctp_max_message_size_from_media (const GstSDPMedia * media)
return 65536;
}
gboolean
_message_media_is_datachannel (const GstSDPMessage * msg, guint media_id)
{
const GstSDPMedia *media;
if (!msg)
return FALSE;
if (gst_sdp_message_medias_len (msg) <= media_id)
return FALSE;
media = gst_sdp_message_get_media (msg, media_id);
if (g_strcmp0 (gst_sdp_media_get_media (media), "application") != 0)
return FALSE;
if (gst_sdp_media_formats_len (media) != 1)
return FALSE;
if (g_strcmp0 (gst_sdp_media_get_format (media, 0),
"webrtc-datachannel") != 0)
return FALSE;
return TRUE;
}
void
_get_ice_credentials_from_sdp_media (const GstSDPMessage * sdp, guint media_idx,
gchar ** ufrag, gchar ** pwd)
{
int i;
*ufrag = NULL;
*pwd = NULL;
{
/* search in the corresponding media section */
const GstSDPMedia *media = gst_sdp_message_get_media (sdp, media_idx);
const gchar *tmp_ufrag =
gst_sdp_media_get_attribute_val (media, "ice-ufrag");
const gchar *tmp_pwd = gst_sdp_media_get_attribute_val (media, "ice-pwd");
if (tmp_ufrag && tmp_pwd) {
*ufrag = g_strdup (tmp_ufrag);
*pwd = g_strdup (tmp_pwd);
return;
}
}
/* then in the sdp message itself */
for (i = 0; i < gst_sdp_message_attributes_len (sdp); i++) {
const GstSDPAttribute *attr = gst_sdp_message_get_attribute (sdp, i);
if (g_strcmp0 (attr->key, "ice-ufrag") == 0) {
g_assert (!*ufrag);
*ufrag = g_strdup (attr->value);
} else if (g_strcmp0 (attr->key, "ice-pwd") == 0) {
g_assert (!*pwd);
*pwd = g_strdup (attr->value);
}
}
if (!*ufrag && !*pwd) {
/* Check in the medias themselves. According to JSEP, they should be
* identical FIXME: only for bundle-d streams */
for (i = 0; i < gst_sdp_message_medias_len (sdp); i++) {
const GstSDPMedia *media = gst_sdp_message_get_media (sdp, i);
const gchar *tmp_ufrag =
gst_sdp_media_get_attribute_val (media, "ice-ufrag");
const gchar *tmp_pwd = gst_sdp_media_get_attribute_val (media, "ice-pwd");
if (tmp_ufrag && tmp_pwd) {
*ufrag = g_strdup (tmp_ufrag);
*pwd = g_strdup (tmp_pwd);
break;
}
}
}
}
gboolean
_parse_bundle (GstSDPMessage * sdp, GStrv * bundled)
{
const gchar *group;
gboolean ret = FALSE;
group = gst_sdp_message_get_attribute_val (sdp, "group");
if (group && g_str_has_prefix (group, "BUNDLE ")) {
*bundled = g_strsplit (group + strlen ("BUNDLE "), " ", 0);
if (!(*bundled)[0]) {
GST_ERROR ("Invalid format for BUNDLE group, expected at least "
"one mid (%s)", group);
goto done;
}
} else {
ret = TRUE;
goto done;
}
ret = TRUE;
done:
return ret;
}
gboolean
_get_bundle_index (GstSDPMessage * sdp, GStrv bundled, guint * idx)
{
gboolean ret = FALSE;
guint i;
for (i = 0; i < gst_sdp_message_medias_len (sdp); i++) {
const GstSDPMedia *media = gst_sdp_message_get_media (sdp, i);
const gchar *mid = gst_sdp_media_get_attribute_val (media, "mid");
if (!g_strcmp0 (mid, bundled[0])) {
*idx = i;
ret = TRUE;
break;
}
}
return ret;
}

17
ext/webrtc/webrtcsdp.h
View file

@ -81,4 +81,21 @@ int _get_sctp_port_from_media (con
G_GNUC_INTERNAL
guint64 _get_sctp_max_message_size_from_media (const GstSDPMedia * media);
G_GNUC_INTERNAL
void _get_ice_credentials_from_sdp_media (const GstSDPMessage * sdp,
guint media_idx,
gchar ** ufrag,
gchar ** pwd);
G_GNUC_INTERNAL
gboolean _message_media_is_datachannel (const GstSDPMessage * msg,
guint media_id);
G_GNUC_INTERNAL
gboolean _get_bundle_index (GstSDPMessage * sdp,
GStrv bundled,
guint * idx);
G_GNUC_INTERNAL
gboolean _parse_bundle (GstSDPMessage * sdp,
GStrv * bundled);
#endif /* __WEBRTC_UTILS_H__ */

28
ext/webrtc/webrtctransceiver.c
View file

@ -69,6 +69,34 @@ webrtc_transceiver_set_transport (WebRTCTransceiver * trans,
(GstObject *) stream->rtcp_transport);
}
GstWebRTCDTLSTransport *
webrtc_transceiver_get_dtls_transport (GstWebRTCRTPTransceiver * trans)
{
g_return_val_if_fail (WEBRTC_IS_TRANSCEIVER (trans), NULL);
if (trans->sender) {
return trans->sender->transport;
} else if (trans->receiver) {
return trans->receiver->transport;
}
return NULL;
}
GstWebRTCDTLSTransport *
webrtc_transceiver_get_rtcp_dtls_transport (GstWebRTCRTPTransceiver * trans)
{
g_return_val_if_fail (WEBRTC_IS_TRANSCEIVER (trans), NULL);
if (trans->sender) {
return trans->sender->rtcp_transport;
} else if (trans->receiver) {
return trans->receiver->rtcp_transport;
}
return NULL;
}
static void
webrtc_transceiver_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)

3
ext/webrtc/webrtctransceiver.h
View file

@ -58,6 +58,9 @@ WebRTCTransceiver * webrtc_transceiver_new (GstWebRTCBin * webr
void webrtc_transceiver_set_transport (WebRTCTransceiver * trans,
TransportStream * stream);
GstWebRTCDTLSTransport * webrtc_transceiver_get_dtls_transport (GstWebRTCRTPTransceiver * trans);
GstWebRTCDTLSTransport * webrtc_transceiver_get_rtcp_dtls_transport (GstWebRTCRTPTransceiver * trans);
G_END_DECLS
#endif /* __WEBRTC_TRANSCEIVER_H__ */