- @FREETYPE2_CFLAGS@ \
- @WAYLAND_CFLAGS@ \
- @GL_CFLAGS@ \
-+ @GSTREAMER_CFLAGS@ \
-+ @LIBAFBWSC_CFLAGS@ \
- @ZLIB_CFLAGS@ \
- @SQLITE3_CFLAGS@ \
- @EXPAT_CFLAGS@ \
-@@ -555,6 +560,8 @@ libnavicore_la_CXXFLAGS = -fPIC \
- libnavicore_la_LIBADD = \
- @OPENSSL_LIBS@ \
- @GL_LIBS@ \
-+ @GSTREAMER_LIBS@ \
-+ @LIBAFBWSC_LIBS@ \
- @ZLIB_LIBS@ \
- @SQLITE3_LIBS@ \
- @EXPAT_LIBS@
-diff --git a/src/sms/sms-core/SMCoreDM/RG/RG_GuideVoice.c b/src/sms/sms-core/SMCoreDM/RG/RG_GuideVoice.c
-index 3828d5c..36e6775 100755
---- a/src/sms/sms-core/SMCoreDM/RG/RG_GuideVoice.c
-+++ b/src/sms/sms-core/SMCoreDM/RG/RG_GuideVoice.c
-@@ -16,6 +16,8 @@
-
- #include "sms-core/SMCoreDM/SMCoreDMInternal.h"
-
-+extern int play_voice(const char* voice_gen_cmd);
-+
- typedef struct _tts_text_tbl
- {
- INT32 code;
-@@ -205,9 +207,9 @@ E_SC_RESULT RG_CTL_CreateVoiceText(RT_NAME_t *in, INT32 language)
- }
- else
- {
-- strncat(tts_voice, "\" & ", (TTSMAX - len - 1));
-+ strncat(tts_voice, "\"", (TTSMAX - len - 1));
-
-- system(tts_voice);
-+ play_voice(tts_voice);
- }
-
- return (e_SC_RESULT_SUCCESS);
-diff --git a/src/sms/sms-core/SMCoreDM/RG/RG_GuideVoice4A.cpp b/src/sms/sms-core/SMCoreDM/RG/RG_GuideVoice4A.cpp
-new file mode 100644
-index 0000000..6b59c0e
---- /dev/null
-+++ b/src/sms/sms-core/SMCoreDM/RG/RG_GuideVoice4A.cpp
-@@ -0,0 +1,223 @@
-+/*
-+ * Copyright (C) 2018 Konsulko Group
-+ * Author: Scott Murray <scott.murray@konsulko.com>
-+ *
-+ * This program is licensed under GPL version 2 license.
-+ * See the LICENSE file distributed with this source file.
-+ */
-+
-+#include <string>
-+#include <iostream>
-+#include <cstring>
-+#include <unistd.h>
-+#include <sys/types.h>
-+#include <sys/stat.h>
-+#include <pthread.h>
-+#include <gst/gst.h>
-+#include <json-c/json.h>
-+
-+extern "C"
-+{
-+#include <afb/afb-wsj1.h>
-+#include <afb/afb-ws-client.h>
-+#include <systemd/sd-event.h>
-+}
-+
-+#define NAVI_TMPFILE "/tmp/navi.wav"
-+
-+static struct afb_wsj1* ws;
-+static struct afb_wsj1_itf itf;
-+sd_event* loop;
-+
-+// port and token from src/glview/glview_wayland.cpp
-+extern long g_port;
-+extern std::string g_token;
-+
-+static int set_role_state(bool state);
-+
-+void play_voice_file(const char *output)
-+{
-+ if(!output)
-+ return;
-+
-+ // Initialize GStreamer
-+ gst_init(NULL, NULL);
-+
-+ std::string pipeline_str = "filesrc location=";
-+ pipeline_str += NAVI_TMPFILE;
-+ pipeline_str += " ! wavparse ! audioconvert ! audioresample ! alsasink device=";
-+ pipeline_str += output;
-+ GstElement *pipeline = gst_parse_launch(pipeline_str.c_str(), NULL);
-+ if(!pipeline) {
-+ std::cerr << "gstreamer pipeline construction failed!" << std::endl;
-+ return;
-+ }
-+
-+ // Start pipeline
-+ gst_element_set_state(pipeline, GST_STATE_PLAYING);
-+ std::cerr << "Playing guidance" << std::endl;
-+
-+ // Wait until error or EOS
-+ GstBus *bus = gst_element_get_bus(pipeline);
-+ GstMessage *msg = gst_bus_timed_pop_filtered(bus,
-+ GST_CLOCK_TIME_NONE,
-+ (GstMessageType) (GST_MESSAGE_ERROR | GST_MESSAGE_EOS));
-+
-+ // Free resources
-+ if(msg != NULL)
-+ gst_message_unref(msg);
-+ gst_object_unref(bus);
-+ gst_element_set_state(pipeline, GST_STATE_NULL);
-+ gst_object_unref(pipeline);
-+
-+ // Remove temporary file
-+ unlink(NAVI_TMPFILE);
-+
-+ return;
-+}
-+
-+static void on_hangup(void *closure, struct afb_wsj1 *wsj)
-+{
-+}
-+
-+static void on_call(void *closure, const char *api, const char *verb, struct afb_wsj1_msg *msg)
-+{
-+}
-+
-+static void on_event(void* closure, const char* event, struct afb_wsj1_msg *msg)
-+{
-+}
-+
-+static void on_reply(void *closure, struct afb_wsj1_msg *msg)
-+{
-+ bool state = (bool) closure;
-+
-+ if(!state) {
-+ // Role is closed, return
-+ return;
-+ }
-+
-+ // We opened the role, play the file
-+ struct json_object* reply = afb_wsj1_msg_object_j(msg);
-+ if(reply) {
-+ struct json_object* response;
-+ int rc = json_object_object_get_ex(reply, "response", &response);
-+ if(rc) {
-+ struct json_object* val;
-+ rc = json_object_object_get_ex(response, "device_uri", &val);
-+ if (rc && json_object_get_string_len(val)) {
-+ const char* jres_pcm = json_object_get_string(val);
-+ play_voice_file(jres_pcm);
-+ }
-+ }
-+ }
-+
-+ // Give up role now that we're done
-+ set_role_state(false);
-+}
-+
-+static void *event_loop_run(void *args)
-+{
-+ struct sd_event* loop = (struct sd_event*)(args);
-+
-+ for(;;)
-+ sd_event_run(loop, 30000000);
-+}
-+
-+static int start_event_loop(void)
-+{
-+ if(ws && loop) {
-+ pthread_t thread_id;
-+ if(pthread_create(&thread_id, NULL, event_loop_run, loop) != 0) {
-+ return -1;
-+ } else {
-+ return thread_id;
-+ }
-+ } else {
-+ return -1;
-+ }
-+}
-+
-+static int init_ws(int port, std::string &token)
-+{
-+ loop = NULL;
-+ std::string uri;
-+
-+ if(sd_event_default(&loop) < 0) {
-+ std::cerr << __FUNCTION__ << ": Failed to create event loop" << std::endl;
-+ goto error;
-+ }
-+
-+ // Initialize interface for websocket
-+ itf.on_hangup = on_hangup;
-+ itf.on_call = on_call;
-+ itf.on_event = on_event;
-+
-+ uri = "ws://localhost:" + std::to_string(port) + "/api?token=" + token;
-+ std::cerr << "Using URI: " << uri << std::endl;
-+ ws = afb_ws_client_connect_wsj1(loop, uri.c_str(), &itf, NULL);
-+ if(ws == NULL) {
-+ std::cerr << __FUNCTION__ << ": Failed to create websocket connection" << std::endl;
-+ goto error;
-+ }
-+ start_event_loop();
-+ return 0;
-+error:
-+ if(loop) {
-+ sd_event_unref(loop);
-+ }
-+ return -1;
-+}
-+
-+static int set_role_state(bool state)
-+{
-+ int rc;
-+ json_object *jsonData = json_object_new_object();
-+
-+ json_object_object_add(jsonData, "action", json_object_new_string(state ? "open" : "close"));
-+ rc = afb_wsj1_call_j(ws, "ahl-4a", "navigation", jsonData, on_reply, (void*) state);
-+ if (rc < 0) {
-+ std::cerr << __FUNCTION__ << ": Failed to call ahl-4a/navigation!" << std::endl;
-+ }
-+ return rc;
-+}
-+
-+pthread_mutex_t ws_mutex = PTHREAD_MUTEX_INITIALIZER;
-+
-+static void *play_voice_handler(void *data)
-+{
-+ int rc;
-+ char *voice_gen_cmd = (char*) data;
-+ if(!voice_gen_cmd)
-+ return NULL;
-+
-+ pthread_mutex_lock(&ws_mutex);
-+ if(!ws) {
-+ rc = init_ws(g_port, g_token);
-+ pthread_mutex_unlock(&ws_mutex);
-+ if(rc < 0)
-+ return NULL;
-+ }
-+ pthread_mutex_unlock(&ws_mutex);
-+
-+ // Generate guidance voice file
-+ rc = system(voice_gen_cmd);
-+ free(voice_gen_cmd);
-+
-+ // Try to get role and play file
-+ set_role_state(true);
-+
-+ return NULL;
-+}
-+
-+extern "C" int play_voice(const char* voice_gen_cmd)
-+{
-+ pthread_t handler_thread;
-+ char *tmp;
-+
-+ if(!voice_gen_cmd)
-+ return -1;
-+
-+ tmp = strdup(voice_gen_cmd);
-+ return pthread_create(&handler_thread, NULL, play_voice_handler, (void*) tmp);
-+}
@@ -0,0 +1,35 @@
+gpsnavi: Switch to ALSA output
+
+Update the talk scripts to use ALSA output via gst-launch-1.0 instead
+of PulseAudio's paplay. gstreamer is used since it is likely that a
+further revision will change to a pipewire output sink and add back
+setting a role property.
+
+Upstream-Status: Inappropriate [no upstream]
+
+Signed-off-by: Scott Murray <scott.murray@konsulko.com>
+
+diff --git a/flite_agl.in b/flite_agl.in
+index 28b512c..67a09e5 100644
+--- a/flite_agl.in
++++ b/flite_agl.in
+@@ -1,6 +1,6 @@
+ #!/bin/sh
+ TMP=/tmp/navi.wav
+ echo "$1" | flite_hts_engine -m @datadir@/Voice/us/cmu_us_arctic_slt.htsvoice -o $TMP
+-paplay --property='media.role=Navi' $TMP
++gst-launch-1.0 filesrc location=$TMP ! decodebin ! audioconvert ! audioresample ! alsasink
+ rm -f $TMP
+
+diff --git a/jtalk_agl.in b/jtalk_agl.in
+index 76900f4..73c87e5 100644
+--- a/jtalk_agl.in
++++ b/jtalk_agl.in
+@@ -1,6 +1,6 @@
+ #!/bin/sh
+ TMP=/tmp/navi.wav
+ echo "$1" | open_jtalk -ow $TMP -m @exec_prefix@/share/Voice/mei/mei_normal.htsvoice -x @exec_prefix@/share/dic/
+-paplay --property='media.role=Navi' $TMP
++gst-launch-1.0 filesrc location=$TMP ! decodebin ! audioconvert ! audioresample ! alsasink
+ rm -f $TMP
+
@@ -0,0 +1,22 @@
+gpsnavi: Update permissions
+
+Add the new display and audio permissions required with the change to
+running as non-root.
+
+Upstream-Status: Inappropriate [no upstream]
+
+Signed-off-by: Scott Murray <scott.murray@konsulko.com>
+
+diff --git a/agl/config.xml b/agl/config.xml
+index 9d4c0ca..44de94a 100755
+--- a/agl/config.xml
++++ b/agl/config.xml
+@@ -8,6 +8,8 @@
+ <feature name="urn:AGL:widget:required-permission">
+ <param name="urn:AGL:permission::public:no-htdocs" value="required" />
+ <param name="http://tizen.org/privilege/internal/dbus" value="required" />
++ <param name="urn:AGL:permission::public:display" value="required" />
++ <param name="urn:AGL:permission::public:audio" value="required" />
+ </feature>
+ <feature name="urn:AGL:widget:required-api">
+ <param name="homescreen" value="ws" />
@@ -13,14 +13,15 @@ DEPENDS = " \
"
RDEPENDS_${PN} = " flite openjtalk glib-2.0 freetype sqlite3 wayland zlib expat openssl \
- wayland libdbus-c++ af-main "
+ wayland libdbus-c++ af-main gstreamer1.0"
RDEPENDS_${PN} += " agl-service-navigation "
SRCREV="89dc0052aced411ef09f8e0034fb5cf2c96ee637"
SRC_URI="git://github.com/AGLExport/gpsnavi.git;branch=agl \
- file://0001-add-4A-playback-support.patch \
+ file://0001-switch-to-alsa-output.patch \
file://0002-openssl-1.1-fixes.patch \
+ file://0003-update-permissions.patch \
file://download_mapdata_jp.sh \
file://download_mapdata_uk.sh \
file://org.agl.naviapi.conf \
Scott Murray <scott.murray@konsulko.com>