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webrtc: indent sources

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Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-examples/-/merge_requests/16>
This commit is contained in:
Matthew Waters 2020-06-19 12:30:23 +10:00
parent e1c3dad258
commit 204945b902
3 changed files with 140 additions and 123 deletions
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228
webrtc/android/app/src/main/jni/webrtc.c
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@ -43,24 +43,25 @@ GST_DEBUG_CATEGORY_STATIC (debug_category);
#define GET_CUSTOM_DATA(env, thiz, fieldID) (WebRTC *)(gintptr)(*env)->GetLongField (env, thiz, fieldID) #define GET_CUSTOM_DATA(env, thiz, fieldID) (WebRTC *)(gintptr)(*env)->GetLongField (env, thiz, fieldID)
#define SET_CUSTOM_DATA(env, thiz, fieldID, data) (*env)->SetLongField (env, thiz, fieldID, (jlong)(gintptr)data) #define SET_CUSTOM_DATA(env, thiz, fieldID, data) (*env)->SetLongField (env, thiz, fieldID, (jlong)(gintptr)data)
enum AppState { enum AppState
APP_STATE_UNKNOWN = 0, {
APP_STATE_ERROR = 1, /* generic error */ APP_STATE_UNKNOWN = 0,
SERVER_CONNECTING = 1000, APP_STATE_ERROR = 1, /* generic error */
SERVER_CONNECTION_ERROR, SERVER_CONNECTING = 1000,
SERVER_CONNECTED, /* Ready to register */ SERVER_CONNECTION_ERROR,
SERVER_REGISTERING = 2000, SERVER_CONNECTED, /* Ready to register */
SERVER_REGISTRATION_ERROR, SERVER_REGISTERING = 2000,
SERVER_REGISTERED, /* Ready to call a peer */ SERVER_REGISTRATION_ERROR,
SERVER_CLOSED, /* server connection closed by us or the server */ SERVER_REGISTERED, /* Ready to call a peer */
PEER_CONNECTING = 3000, SERVER_CLOSED, /* server connection closed by us or the server */
PEER_CONNECTION_ERROR, PEER_CONNECTING = 3000,
PEER_CONNECTED, PEER_CONNECTION_ERROR,
PEER_CALL_NEGOTIATING = 4000, PEER_CONNECTED,
PEER_CALL_STARTED, PEER_CALL_NEGOTIATING = 4000,
PEER_CALL_STOPPING, PEER_CALL_STARTED,
PEER_CALL_STOPPED, PEER_CALL_STOPPING,
PEER_CALL_ERROR, PEER_CALL_STOPPED,
PEER_CALL_ERROR,
}; };
typedef struct _WebRTC typedef struct _WebRTC
@ -115,7 +116,7 @@ cleanup_and_quit_loop (WebRTC * webrtc, const gchar * msg, enum AppState state)
return G_SOURCE_REMOVE; return G_SOURCE_REMOVE;
} }
static gchar* static gchar *
get_string_from_json_object (JsonObject * object) get_string_from_json_object (JsonObject * object)
{ {
JsonNode *root; JsonNode *root;
@ -135,8 +136,8 @@ get_string_from_json_object (JsonObject * object)
} }
static GstElement * static GstElement *
handle_media_stream (GstPad * pad, GstElement * pipe, const char * convert_name, handle_media_stream (GstPad * pad, GstElement * pipe, const char *convert_name,
const char * sink_name) const char *sink_name)
{ {
GstPad *qpad; GstPad *qpad;
GstElement *q, *conv, *sink; GstElement *q, *conv, *sink;
@ -176,14 +177,14 @@ handle_media_stream (GstPad * pad, GstElement * pipe, const char * convert_name,
static void static void
on_incoming_decodebin_stream (GstElement * decodebin, GstPad * pad, on_incoming_decodebin_stream (GstElement * decodebin, GstPad * pad,
WebRTC * webrtc) WebRTC * webrtc)
{ {
GstCaps *caps; GstCaps *caps;
const gchar *name; const gchar *name;
if (!gst_pad_has_current_caps (pad)) { if (!gst_pad_has_current_caps (pad)) {
g_printerr ("Pad '%s' has no caps, can't do anything, ignoring\n", g_printerr ("Pad '%s' has no caps, can't do anything, ignoring\n",
GST_PAD_NAME (pad)); GST_PAD_NAME (pad));
return; return;
} }
@ -191,11 +192,13 @@ on_incoming_decodebin_stream (GstElement * decodebin, GstPad * pad,
name = gst_structure_get_name (gst_caps_get_structure (caps, 0)); name = gst_structure_get_name (gst_caps_get_structure (caps, 0));
if (g_str_has_prefix (name, "video")) { if (g_str_has_prefix (name, "video")) {
GstElement *sink = handle_media_stream (pad, webrtc->pipe, "videoconvert", "glimagesink"); GstElement *sink =
handle_media_stream (pad, webrtc->pipe, "videoconvert", "glimagesink");
if (webrtc->video_sink == NULL) { if (webrtc->video_sink == NULL) {
webrtc->video_sink = sink; webrtc->video_sink = sink;
if (webrtc->native_window) if (webrtc->native_window)
gst_video_overlay_set_window_handle (GST_VIDEO_OVERLAY (sink), (gpointer) webrtc->native_window); gst_video_overlay_set_window_handle (GST_VIDEO_OVERLAY (sink),
(gpointer) webrtc->native_window);
} }
} else if (g_str_has_prefix (name, "audio")) { } else if (g_str_has_prefix (name, "audio")) {
handle_media_stream (pad, webrtc->pipe, "audioconvert", "autoaudiosink"); handle_media_stream (pad, webrtc->pipe, "audioconvert", "autoaudiosink");
@ -216,21 +219,22 @@ on_incoming_stream (GstElement * webrtcbin, GstPad * pad, WebRTC * webrtc)
decodebin = gst_element_factory_make ("decodebin", NULL); decodebin = gst_element_factory_make ("decodebin", NULL);
g_signal_connect (decodebin, "pad-added", g_signal_connect (decodebin, "pad-added",
G_CALLBACK (on_incoming_decodebin_stream), webrtc); G_CALLBACK (on_incoming_decodebin_stream), webrtc);
gst_bin_add (GST_BIN (webrtc->pipe), decodebin); gst_bin_add (GST_BIN (webrtc->pipe), decodebin);
gst_element_sync_state_with_parent (decodebin); gst_element_sync_state_with_parent (decodebin);
gst_element_link (webrtcbin, decodebin); gst_element_link (webrtcbin, decodebin);
} }
static void static void
send_ice_candidate_message (GstElement * webrtcbin G_GNUC_UNUSED, guint mlineindex, send_ice_candidate_message (GstElement * webrtcbin G_GNUC_UNUSED,
gchar * candidate, WebRTC * webrtc) guint mlineindex, gchar * candidate, WebRTC * webrtc)
{ {
gchar *text; gchar *text;
JsonObject *ice, *msg; JsonObject *ice, *msg;
if (webrtc->app_state < PEER_CALL_NEGOTIATING) { if (webrtc->app_state < PEER_CALL_NEGOTIATING) {
cleanup_and_quit_loop (webrtc, "Can't send ICE, not in call", APP_STATE_ERROR); cleanup_and_quit_loop (webrtc, "Can't send ICE, not in call",
APP_STATE_ERROR);
return; return;
} }
@ -253,7 +257,8 @@ send_sdp_offer (WebRTC * webrtc, GstWebRTCSessionDescription * offer)
JsonObject *msg, *sdp; JsonObject *msg, *sdp;
if (webrtc->app_state < PEER_CALL_NEGOTIATING) { if (webrtc->app_state < PEER_CALL_NEGOTIATING) {
cleanup_and_quit_loop (webrtc, "Can't send offer, not in call", APP_STATE_ERROR); cleanup_and_quit_loop (webrtc, "Can't send offer, not in call",
APP_STATE_ERROR);
return; return;
} }
@ -283,14 +288,15 @@ on_offer_created (GstPromise * promise, WebRTC * webrtc)
g_assert (webrtc->app_state == PEER_CALL_NEGOTIATING); g_assert (webrtc->app_state == PEER_CALL_NEGOTIATING);
g_assert (gst_promise_wait(promise) == GST_PROMISE_RESULT_REPLIED); g_assert (gst_promise_wait (promise) == GST_PROMISE_RESULT_REPLIED);
reply = gst_promise_get_reply (promise); reply = gst_promise_get_reply (promise);
gst_structure_get (reply, "offer", gst_structure_get (reply, "offer",
GST_TYPE_WEBRTC_SESSION_DESCRIPTION, &offer, NULL); GST_TYPE_WEBRTC_SESSION_DESCRIPTION, &offer, NULL);
gst_promise_unref (promise); gst_promise_unref (promise);
promise = gst_promise_new (); promise = gst_promise_new ();
g_signal_emit_by_name (webrtc->webrtcbin, "set-local-description", offer, promise); g_signal_emit_by_name (webrtc->webrtcbin, "set-local-description", offer,
promise);
gst_promise_interrupt (promise); gst_promise_interrupt (promise);
gst_promise_unref (promise); gst_promise_unref (promise);
@ -333,13 +339,12 @@ start_pipeline (WebRTC * webrtc)
GstPad *pad; GstPad *pad;
webrtc->pipe = webrtc->pipe =
gst_parse_launch ("webrtcbin name=sendrecv " gst_parse_launch ("webrtcbin name=sendrecv "
"ahcsrc device-facing=front ! video/x-raw,width=[320,1280] ! queue max-size-buffers=1 ! videoconvert ! " "ahcsrc device-facing=front ! video/x-raw,width=[320,1280] ! queue max-size-buffers=1 ! videoconvert ! "
"vp8enc keyframe-max-dist=30 deadline=1 error-resilient=default ! rtpvp8pay picture-id-mode=15-bit mtu=1300 ! " "vp8enc keyframe-max-dist=30 deadline=1 error-resilient=default ! rtpvp8pay picture-id-mode=15-bit mtu=1300 ! "
"queue max-size-time=300000000 ! " RTP_CAPS_VP8 " ! sendrecv.sink_0 " "queue max-size-time=300000000 ! " RTP_CAPS_VP8 " ! sendrecv.sink_0 "
"openslessrc ! queue ! audioconvert ! audioresample ! audiorate ! queue ! opusenc ! rtpopuspay ! " "openslessrc ! queue ! audioconvert ! audioresample ! audiorate ! queue ! opusenc ! rtpopuspay ! "
"queue ! " RTP_CAPS_OPUS " ! sendrecv.sink_1 ", "queue ! " RTP_CAPS_OPUS " ! sendrecv.sink_1 ", &error);
&error);
if (error) { if (error) {
g_printerr ("Failed to parse launch: %s\n", error->message); g_printerr ("Failed to parse launch: %s\n", error->message);
@ -354,15 +359,15 @@ start_pipeline (WebRTC * webrtc)
/* This is the gstwebrtc entry point where we create the offer and so on. It /* This is the gstwebrtc entry point where we create the offer and so on. It
* will be called when the pipeline goes to PLAYING. */ * will be called when the pipeline goes to PLAYING. */
g_signal_connect (webrtc->webrtcbin, "on-negotiation-needed", g_signal_connect (webrtc->webrtcbin, "on-negotiation-needed",
G_CALLBACK (on_negotiation_needed), webrtc); G_CALLBACK (on_negotiation_needed), webrtc);
/* We need to transmit this ICE candidate to the browser via the websockets /* We need to transmit this ICE candidate to the browser via the websockets
* signalling server. Incoming ice candidates from the browser need to be * signalling server. Incoming ice candidates from the browser need to be
* added by us too, see on_server_message() */ * added by us too, see on_server_message() */
g_signal_connect (webrtc->webrtcbin, "on-ice-candidate", g_signal_connect (webrtc->webrtcbin, "on-ice-candidate",
G_CALLBACK (send_ice_candidate_message), webrtc); G_CALLBACK (send_ice_candidate_message), webrtc);
/* Incoming streams will be exposed via this signal */ /* Incoming streams will be exposed via this signal */
g_signal_connect (webrtc->webrtcbin, "pad-added", G_CALLBACK (on_incoming_stream), g_signal_connect (webrtc->webrtcbin, "pad-added",
webrtc); G_CALLBACK (on_incoming_stream), webrtc);
/* Lifetime is the same as the pipeline itself */ /* Lifetime is the same as the pipeline itself */
gst_object_unref (webrtc->webrtcbin); gst_object_unref (webrtc->webrtcbin);
@ -425,8 +430,7 @@ register_with_server (WebRTC * webrtc)
} }
static void static void
on_server_closed (SoupWebsocketConnection * conn G_GNUC_UNUSED, on_server_closed (SoupWebsocketConnection * conn G_GNUC_UNUSED, WebRTC * webrtc)
WebRTC * webrtc)
{ {
webrtc->app_state = SERVER_CLOSED; webrtc->app_state = SERVER_CLOSED;
cleanup_and_quit_loop (webrtc, "Server connection closed", 0); cleanup_and_quit_loop (webrtc, "Server connection closed", 0);
@ -435,7 +439,7 @@ on_server_closed (SoupWebsocketConnection * conn G_GNUC_UNUSED,
/* One mega message handler for our asynchronous calling mechanism */ /* One mega message handler for our asynchronous calling mechanism */
static void static void
on_server_message (SoupWebsocketConnection * conn, SoupWebsocketDataType type, on_server_message (SoupWebsocketConnection * conn, SoupWebsocketDataType type,
GBytes * message, WebRTC * webrtc) GBytes * message, WebRTC * webrtc)
{ {
gsize size; gsize size;
gchar *text, *data; gchar *text, *data;
@ -443,14 +447,14 @@ on_server_message (SoupWebsocketConnection * conn, SoupWebsocketDataType type,
switch (type) { switch (type) {
case SOUP_WEBSOCKET_DATA_BINARY: case SOUP_WEBSOCKET_DATA_BINARY:
g_printerr ("Received unknown binary message, ignoring\n"); g_printerr ("Received unknown binary message, ignoring\n");
g_bytes_unref (message); g_bytes_unref (message);
return; return;
case SOUP_WEBSOCKET_DATA_TEXT: case SOUP_WEBSOCKET_DATA_TEXT:
data = g_bytes_unref_to_data (message, &size); data = g_bytes_unref_to_data (message, &size);
/* Convert to NULL-terminated string */ /* Convert to NULL-terminated string */
text = g_strndup (data, size); text = g_strndup (data, size);
g_free (data); g_free (data);
break; break;
default: default:
g_assert_not_reached (); g_assert_not_reached ();
} }
@ -458,22 +462,23 @@ on_server_message (SoupWebsocketConnection * conn, SoupWebsocketDataType type,
/* Server has accepted our registration, we are ready to send commands */ /* Server has accepted our registration, we are ready to send commands */
if (g_strcmp0 (text, "HELLO") == 0) { if (g_strcmp0 (text, "HELLO") == 0) {
if (webrtc->app_state != SERVER_REGISTERING) { if (webrtc->app_state != SERVER_REGISTERING) {
cleanup_and_quit_loop (webrtc, "ERROR: Received HELLO when not registering", cleanup_and_quit_loop (webrtc,
APP_STATE_ERROR); "ERROR: Received HELLO when not registering", APP_STATE_ERROR);
goto out; goto out;
} }
webrtc->app_state = SERVER_REGISTERED; webrtc->app_state = SERVER_REGISTERED;
g_print ("Registered with server\n"); g_print ("Registered with server\n");
/* Ask signalling server to connect us with a specific peer */ /* Ask signalling server to connect us with a specific peer */
if (!setup_call (webrtc)) { if (!setup_call (webrtc)) {
cleanup_and_quit_loop (webrtc, "ERROR: Failed to setup call", PEER_CALL_ERROR); cleanup_and_quit_loop (webrtc, "ERROR: Failed to setup call",
PEER_CALL_ERROR);
goto out; goto out;
} }
/* Call has been setup by the server, now we can start negotiation */ /* Call has been setup by the server, now we can start negotiation */
} else if (g_strcmp0 (text, "SESSION_OK") == 0) { } else if (g_strcmp0 (text, "SESSION_OK") == 0) {
if (webrtc->app_state != PEER_CONNECTING) { if (webrtc->app_state != PEER_CONNECTING) {
cleanup_and_quit_loop (webrtc, "ERROR: Received SESSION_OK when not calling", cleanup_and_quit_loop (webrtc,
PEER_CONNECTION_ERROR); "ERROR: Received SESSION_OK when not calling", PEER_CONNECTION_ERROR);
goto out; goto out;
} }
@ -481,23 +486,23 @@ on_server_message (SoupWebsocketConnection * conn, SoupWebsocketDataType type,
/* Start negotiation (exchange SDP and ICE candidates) */ /* Start negotiation (exchange SDP and ICE candidates) */
if (!start_pipeline (webrtc)) if (!start_pipeline (webrtc))
cleanup_and_quit_loop (webrtc, "ERROR: failed to start pipeline", cleanup_and_quit_loop (webrtc, "ERROR: failed to start pipeline",
PEER_CALL_ERROR); PEER_CALL_ERROR);
/* Handle errors */ /* Handle errors */
} else if (g_str_has_prefix (text, "ERROR")) { } else if (g_str_has_prefix (text, "ERROR")) {
switch (webrtc->app_state) { switch (webrtc->app_state) {
case SERVER_CONNECTING: case SERVER_CONNECTING:
webrtc->app_state = SERVER_CONNECTION_ERROR; webrtc->app_state = SERVER_CONNECTION_ERROR;
break; break;
case SERVER_REGISTERING: case SERVER_REGISTERING:
webrtc->app_state = SERVER_REGISTRATION_ERROR; webrtc->app_state = SERVER_REGISTRATION_ERROR;
break; break;
case PEER_CONNECTING: case PEER_CONNECTING:
webrtc->app_state = PEER_CONNECTION_ERROR; webrtc->app_state = PEER_CONNECTION_ERROR;
break; break;
case PEER_CONNECTED: case PEER_CONNECTED:
case PEER_CALL_NEGOTIATING: case PEER_CALL_NEGOTIATING:
webrtc->app_state = PEER_CALL_ERROR; webrtc->app_state = PEER_CALL_ERROR;
break; break;
default: default:
webrtc->app_state = APP_STATE_ERROR; webrtc->app_state = APP_STATE_ERROR;
} }
@ -541,7 +546,7 @@ on_server_message (SoupWebsocketConnection * conn, SoupWebsocketDataType type,
* See tests/examples/webrtcbidirectional.c in gst-plugins-bad for how to * See tests/examples/webrtcbidirectional.c in gst-plugins-bad for how to
* handle offers from peers and reply with answers using webrtcbin. */ * handle offers from peers and reply with answers using webrtcbin. */
g_assert_cmpstr (json_object_get_string_member (object, "type"), ==, g_assert_cmpstr (json_object_get_string_member (object, "type"), ==,
"answer"); "answer");
text = json_object_get_string_member (object, "sdp"); text = json_object_get_string_member (object, "sdp");
@ -554,14 +559,14 @@ on_server_message (SoupWebsocketConnection * conn, SoupWebsocketDataType type,
g_assert (ret == GST_SDP_OK); g_assert (ret == GST_SDP_OK);
answer = gst_webrtc_session_description_new (GST_WEBRTC_SDP_TYPE_ANSWER, answer = gst_webrtc_session_description_new (GST_WEBRTC_SDP_TYPE_ANSWER,
sdp); sdp);
g_assert (answer); g_assert (answer);
/* Set remote description on our pipeline */ /* Set remote description on our pipeline */
{ {
GstPromise *promise = gst_promise_new (); GstPromise *promise = gst_promise_new ();
g_signal_emit_by_name (webrtc->webrtcbin, "set-remote-description", answer, g_signal_emit_by_name (webrtc->webrtcbin, "set-remote-description",
promise); answer, promise);
gst_promise_interrupt (promise); gst_promise_interrupt (promise);
gst_promise_unref (promise); gst_promise_unref (promise);
} }
@ -577,25 +582,25 @@ on_server_message (SoupWebsocketConnection * conn, SoupWebsocketDataType type,
sdpmlineindex = json_object_get_int_member (ice, "sdpMLineIndex"); sdpmlineindex = json_object_get_int_member (ice, "sdpMLineIndex");
/* Add ice candidate sent by remote peer */ /* Add ice candidate sent by remote peer */
g_signal_emit_by_name (webrtc->webrtcbin, "add-ice-candidate", sdpmlineindex, g_signal_emit_by_name (webrtc->webrtcbin, "add-ice-candidate",
candidate); sdpmlineindex, candidate);
} else { } else {
g_printerr ("Ignoring unknown JSON message:\n%s\n", text); g_printerr ("Ignoring unknown JSON message:\n%s\n", text);
} }
g_object_unref (parser); g_object_unref (parser);
} }
out: out:
g_free (text); g_free (text);
} }
static void static void
on_server_connected (SoupSession * session, GAsyncResult * res, on_server_connected (SoupSession * session, GAsyncResult * res, WebRTC * webrtc)
WebRTC * webrtc)
{ {
GError *error = NULL; GError *error = NULL;
webrtc->ws_conn = soup_session_websocket_connect_finish (session, res, &error); webrtc->ws_conn =
soup_session_websocket_connect_finish (session, res, &error);
if (error) { if (error) {
cleanup_and_quit_loop (webrtc, error->message, SERVER_CONNECTION_ERROR); cleanup_and_quit_loop (webrtc, error->message, SERVER_CONNECTION_ERROR);
g_error_free (error); g_error_free (error);
@ -607,8 +612,10 @@ on_server_connected (SoupSession * session, GAsyncResult * res,
webrtc->app_state = SERVER_CONNECTED; webrtc->app_state = SERVER_CONNECTED;
g_print ("Connected to signalling server\n"); g_print ("Connected to signalling server\n");
g_signal_connect (webrtc->ws_conn, "closed", G_CALLBACK (on_server_closed), webrtc); g_signal_connect (webrtc->ws_conn, "closed", G_CALLBACK (on_server_closed),
g_signal_connect (webrtc->ws_conn, "message", G_CALLBACK (on_server_message), webrtc); webrtc);
g_signal_connect (webrtc->ws_conn, "message", G_CALLBACK (on_server_message),
webrtc);
/* Register with the server so it knows about us and can accept commands */ /* Register with the server so it knows about us and can accept commands */
register_with_server (webrtc); register_with_server (webrtc);
@ -623,16 +630,16 @@ connect_to_websocket_server_async (WebRTC * webrtc)
SoupLogger *logger; SoupLogger *logger;
SoupMessage *message; SoupMessage *message;
SoupSession *session; SoupSession *session;
const char *https_aliases[] = {"wss", NULL}; const char *https_aliases[] = { "wss", NULL };
const gchar *ca_certs; const gchar *ca_certs;
ca_certs = g_getenv("CA_CERTIFICATES"); ca_certs = g_getenv ("CA_CERTIFICATES");
g_assert (ca_certs != NULL); g_assert (ca_certs != NULL);
g_print ("ca-certificates %s", ca_certs); g_print ("ca-certificates %s", ca_certs);
session = soup_session_new_with_options (SOUP_SESSION_SSL_STRICT, FALSE, session = soup_session_new_with_options (SOUP_SESSION_SSL_STRICT, FALSE,
// SOUP_SESSION_SSL_USE_SYSTEM_CA_FILE, TRUE, // SOUP_SESSION_SSL_USE_SYSTEM_CA_FILE, TRUE,
SOUP_SESSION_SSL_CA_FILE, ca_certs, SOUP_SESSION_SSL_CA_FILE, ca_certs,
SOUP_SESSION_HTTPS_ALIASES, https_aliases, NULL); SOUP_SESSION_HTTPS_ALIASES, https_aliases, NULL);
logger = soup_logger_new (SOUP_LOGGER_LOG_BODY, -1); logger = soup_logger_new (SOUP_LOGGER_LOG_BODY, -1);
soup_session_add_feature (session, SOUP_SESSION_FEATURE (logger)); soup_session_add_feature (session, SOUP_SESSION_FEATURE (logger));
@ -644,7 +651,7 @@ connect_to_websocket_server_async (WebRTC * webrtc)
/* Once connected, we will register */ /* Once connected, we will register */
soup_session_websocket_connect_async (session, message, NULL, NULL, NULL, soup_session_websocket_connect_async (session, message, NULL, NULL, NULL,
(GAsyncReadyCallback) on_server_connected, webrtc); (GAsyncReadyCallback) on_server_connected, webrtc);
webrtc->app_state = SERVER_CONNECTING; webrtc->app_state = SERVER_CONNECTING;
return G_SOURCE_REMOVE; return G_SOURCE_REMOVE;
@ -708,7 +715,7 @@ native_end_call (JNIEnv * env, jobject thiz)
if (webrtc->loop) { if (webrtc->loop) {
GThread *thread = webrtc->thread; GThread *thread = webrtc->thread;
GST_INFO("Ending current call"); GST_INFO ("Ending current call");
cleanup_and_quit_loop (webrtc, NULL, 0); cleanup_and_quit_loop (webrtc, NULL, 0);
webrtc->thread = NULL; webrtc->thread = NULL;
g_mutex_unlock (&webrtc->lock); g_mutex_unlock (&webrtc->lock);
@ -729,14 +736,15 @@ static gpointer
_call_thread (WebRTC * webrtc) _call_thread (WebRTC * webrtc)
{ {
GMainContext *context = NULL; GMainContext *context = NULL;
JNIEnv *env = attach_current_thread(); JNIEnv *env = attach_current_thread ();
g_mutex_lock (&webrtc->lock); g_mutex_lock (&webrtc->lock);
context = g_main_context_new (); context = g_main_context_new ();
webrtc->loop = g_main_loop_new (context, FALSE); webrtc->loop = g_main_loop_new (context, FALSE);
g_main_context_invoke (context, (GSourceFunc) _unlock_mutex, &webrtc->lock); g_main_context_invoke (context, (GSourceFunc) _unlock_mutex, &webrtc->lock);
g_main_context_invoke (context, (GSourceFunc) connect_to_websocket_server_async, webrtc); g_main_context_invoke (context,
(GSourceFunc) connect_to_websocket_server_async, webrtc);
g_main_context_push_thread_default (context); g_main_context_push_thread_default (context);
g_cond_broadcast (&webrtc->cond); g_cond_broadcast (&webrtc->cond);
g_main_loop_run (webrtc->loop); g_main_loop_run (webrtc->loop);
@ -748,7 +756,7 @@ _call_thread (WebRTC * webrtc)
} }
static void static void
native_call_other_party(JNIEnv * env, jobject thiz) native_call_other_party (JNIEnv * env, jobject thiz)
{ {
WebRTC *webrtc = GET_CUSTOM_DATA (env, thiz, native_webrtc_field_id); WebRTC *webrtc = GET_CUSTOM_DATA (env, thiz, native_webrtc_field_id);
@ -758,9 +766,9 @@ native_call_other_party(JNIEnv * env, jobject thiz)
if (webrtc->thread) if (webrtc->thread)
native_end_call (env, thiz); native_end_call (env, thiz);
GST_INFO("calling other party"); GST_INFO ("calling other party");
webrtc->thread = g_thread_new("webrtc", (GThreadFunc) _call_thread, webrtc); webrtc->thread = g_thread_new ("webrtc", (GThreadFunc) _call_thread, webrtc);
g_mutex_lock (&webrtc->lock); g_mutex_lock (&webrtc->lock);
while (!webrtc->loop) while (!webrtc->loop)
g_cond_wait (&webrtc->cond, &webrtc->lock); g_cond_wait (&webrtc->cond, &webrtc->lock);
@ -814,14 +822,13 @@ native_class_init (JNIEnv * env, jclass klass)
__android_log_print (ANDROID_LOG_ERROR, "GstPlayer", "%s", message); __android_log_print (ANDROID_LOG_ERROR, "GstPlayer", "%s", message);
(*env)->ThrowNew (env, exception_class, message); (*env)->ThrowNew (env, exception_class, message);
} }
//gst_debug_set_threshold_from_string ("gl*:7", FALSE); //gst_debug_set_threshold_from_string ("gl*:7", FALSE);
} }
static void static void
native_set_surface (JNIEnv * env, jobject thiz, jobject surface) native_set_surface (JNIEnv * env, jobject thiz, jobject surface)
{ {
WebRTC *webrtc= GET_CUSTOM_DATA (env, thiz, native_webrtc_field_id); WebRTC *webrtc = GET_CUSTOM_DATA (env, thiz, native_webrtc_field_id);
ANativeWindow *new_native_window; ANativeWindow *new_native_window;
if (!webrtc) if (!webrtc)
@ -829,7 +836,7 @@ native_set_surface (JNIEnv * env, jobject thiz, jobject surface)
new_native_window = surface ? ANativeWindow_fromSurface (env, surface) : NULL; new_native_window = surface ? ANativeWindow_fromSurface (env, surface) : NULL;
GST_DEBUG ("Received surface %p (native window %p)", surface, GST_DEBUG ("Received surface %p (native window %p)", surface,
new_native_window); new_native_window);
if (webrtc->native_window) { if (webrtc->native_window) {
ANativeWindow_release (webrtc->native_window); ANativeWindow_release (webrtc->native_window);
@ -837,36 +844,39 @@ native_set_surface (JNIEnv * env, jobject thiz, jobject surface)
webrtc->native_window = new_native_window; webrtc->native_window = new_native_window;
if (webrtc->video_sink) if (webrtc->video_sink)
gst_video_overlay_set_window_handle (GST_VIDEO_OVERLAY (webrtc->video_sink), (guintptr) new_native_window); gst_video_overlay_set_window_handle (GST_VIDEO_OVERLAY (webrtc->video_sink),
(guintptr) new_native_window);
} }
static void static void
native_set_signalling_server (JNIEnv * env, jobject thiz, jstring server) { native_set_signalling_server (JNIEnv * env, jobject thiz, jstring server)
WebRTC *webrtc= GET_CUSTOM_DATA (env, thiz, native_webrtc_field_id); {
const gchar *s;
if (!webrtc)
return;
s = (*env)->GetStringUTFChars(env, server, NULL);
if (webrtc->signalling_server)
g_free (webrtc->signalling_server);
webrtc->signalling_server = g_strdup (s);
(*env)->ReleaseStringUTFChars(env, server, s);
}
static void
native_set_call_id(JNIEnv * env, jobject thiz, jstring peer_id) {
WebRTC *webrtc = GET_CUSTOM_DATA (env, thiz, native_webrtc_field_id); WebRTC *webrtc = GET_CUSTOM_DATA (env, thiz, native_webrtc_field_id);
const gchar *s; const gchar *s;
if (!webrtc) if (!webrtc)
return; return;
s = (*env)->GetStringUTFChars(env, peer_id, NULL); s = (*env)->GetStringUTFChars (env, server, NULL);
if (webrtc->signalling_server)
g_free (webrtc->signalling_server);
webrtc->signalling_server = g_strdup (s);
(*env)->ReleaseStringUTFChars (env, server, s);
}
static void
native_set_call_id (JNIEnv * env, jobject thiz, jstring peer_id)
{
WebRTC *webrtc = GET_CUSTOM_DATA (env, thiz, native_webrtc_field_id);
const gchar *s;
if (!webrtc)
return;
s = (*env)->GetStringUTFChars (env, peer_id, NULL);
g_free (webrtc->peer_id); g_free (webrtc->peer_id);
webrtc->peer_id = g_strdup (s); webrtc->peer_id = g_strdup (s);
(*env)->ReleaseStringUTFChars(env, peer_id, s); (*env)->ReleaseStringUTFChars (env, peer_id, s);
} }
/* List of implemented native methods */ /* List of implemented native methods */

30
webrtc/sendonly/webrtc-recvonly-h264.c
View file

@ -183,8 +183,8 @@ const gchar *html_source = " \n \
"; ";
static void static void
handle_media_stream (GstPad * pad, GstElement * pipe, const char * convert_name, handle_media_stream (GstPad * pad, GstElement * pipe, const char *convert_name,
const char * sink_name) const char *sink_name)
{ {
GstPad *qpad; GstPad *qpad;
GstElement *q, *conv, *resample, *sink; GstElement *q, *conv, *resample, *sink;
@ -250,7 +250,8 @@ on_incoming_decodebin_stream (GstElement * decodebin, GstPad * pad,
} }
static void static void
on_incoming_stream (GstElement * webrtc, GstPad * pad, ReceiverEntry *receiver_entry) on_incoming_stream (GstElement * webrtc, GstPad * pad,
ReceiverEntry * receiver_entry)
{ {
GstElement *decodebin; GstElement *decodebin;
GstPad *sinkpad; GstPad *sinkpad;
@ -287,10 +288,11 @@ create_receiver_entry (SoupWebsocketConnection * connection)
G_CALLBACK (soup_websocket_message_cb), (gpointer) receiver_entry); G_CALLBACK (soup_websocket_message_cb), (gpointer) receiver_entry);
error = NULL; error = NULL;
receiver_entry->pipeline = gst_parse_launch ("webrtcbin name=webrtcbin stun-server=stun://" STUN_SERVER " " receiver_entry->pipeline =
gst_parse_launch ("webrtcbin name=webrtcbin stun-server=stun://"
STUN_SERVER " "
"audiotestsrc is-live=true wave=red-noise ! audioconvert ! audioresample ! queue ! opusenc ! rtpopuspay ! " "audiotestsrc is-live=true wave=red-noise ! audioconvert ! audioresample ! queue ! opusenc ! rtpopuspay ! "
"queue ! " RTP_CAPS_OPUS "97 ! webrtcbin. " "queue ! " RTP_CAPS_OPUS "97 ! webrtcbin. ", &error);
, &error);
if (error != NULL) { if (error != NULL) {
g_error ("Could not create WebRTC pipeline: %s\n", error->message); g_error ("Could not create WebRTC pipeline: %s\n", error->message);
g_error_free (error); g_error_free (error);
@ -302,18 +304,24 @@ create_receiver_entry (SoupWebsocketConnection * connection)
g_assert (receiver_entry->webrtcbin != NULL); g_assert (receiver_entry->webrtcbin != NULL);
/* Incoming streams will be exposed via this signal */ /* Incoming streams will be exposed via this signal */
g_signal_connect (receiver_entry->webrtcbin, "pad-added", G_CALLBACK (on_incoming_stream), g_signal_connect (receiver_entry->webrtcbin, "pad-added",
receiver_entry); G_CALLBACK (on_incoming_stream), receiver_entry);
#if 0 #if 0
GstElement *rtpbin = gst_bin_get_by_name (GST_BIN (receiver_entry->webrtcbin), "rtpbin"); GstElement *rtpbin =
gst_bin_get_by_name (GST_BIN (receiver_entry->webrtcbin), "rtpbin");
g_object_set (rtpbin, "latency", 40, NULL); g_object_set (rtpbin, "latency", 40, NULL);
gst_object_unref (rtpbin); gst_object_unref (rtpbin);
#endif #endif
// Create a 2nd transceiver for the receive only video stream // Create a 2nd transceiver for the receive only video stream
video_caps = gst_caps_from_string ("application/x-rtp,media=video,encoding-name=H264,payload=" RTP_PAYLOAD_TYPE ",clock-rate=90000,packetization-mode=(string)1, profile-level-id=(string)42c016"); video_caps =
g_signal_emit_by_name (receiver_entry->webrtcbin, "add-transceiver", GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_RECVONLY, video_caps, NULL, &trans); gst_caps_from_string
("application/x-rtp,media=video,encoding-name=H264,payload="
RTP_PAYLOAD_TYPE
",clock-rate=90000,packetization-mode=(string)1, profile-level-id=(string)42c016");
g_signal_emit_by_name (receiver_entry->webrtcbin, "add-transceiver",
GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_RECVONLY, video_caps, NULL, &trans);
gst_caps_unref (video_caps); gst_caps_unref (video_caps);
gst_object_unref (trans); gst_object_unref (trans);

5
webrtc/sendrecv/gst/webrtc-sendrecv.c
View file

@ -525,7 +525,7 @@ on_offer_set (GstPromise * promise, gpointer user_data)
} }
static void static void
on_offer_received (GstSDPMessage *sdp) on_offer_received (GstSDPMessage * sdp)
{ {
GstWebRTCSessionDescription *offer = NULL; GstWebRTCSessionDescription *offer = NULL;
GstPromise *promise; GstPromise *promise;
@ -536,8 +536,7 @@ on_offer_received (GstSDPMessage *sdp)
/* Set remote description on our pipeline */ /* Set remote description on our pipeline */
{ {
promise = gst_promise_new_with_change_func (on_offer_set, NULL, NULL); promise = gst_promise_new_with_change_func (on_offer_set, NULL, NULL);
g_signal_emit_by_name (webrtc1, "set-remote-description", offer, g_signal_emit_by_name (webrtc1, "set-remote-description", offer, promise);
promise);
} }
gst_webrtc_session_description_free (offer); gst_webrtc_session_description_free (offer);
} }