--- /dev/null
+Sangchul Lee <sc11.lee@samsung.com>
+Hyunil Park <hyunil46.park@samsung.com>
+
--- /dev/null
+
+CMAKE_MINIMUM_REQUIRED(VERSION 2.6)
+SET(fw_name "capi-media-webrtc")
+
+PROJECT(${fw_name})
+
+SET(CMAKE_INSTALL_PREFIX /usr)
+SET(PREFIX ${CMAKE_INSTALL_PREFIX})
+SET(LIB_INSTALL_DIR "${CMAKE_INSTALL_PREFIX}/lib")
+
+SET(INC_DIR include)
+INCLUDE_DIRECTORIES(${INC_DIR})
+
+SET(dependents "dlog glib-2.0 gstreamer-1.0" )
+SET(pc_dependents "capi-base-common" )
+
+INCLUDE(FindPkgConfig)
+pkg_check_modules(${fw_name} REQUIRED ${dependents})
+FOREACH(flag ${${fw_name}_CFLAGS})
+ SET(EXTRA_CFLAGS "${EXTRA_CFLAGS} ${flag}")
+ENDFOREACH(flag)
+
+SET(CMAKE_C_FLAGS "${CMAKE_C_FLAGS} ${EXTRA_CFLAGS} -fPIC -Wall -Werror")
+SET(CMAKE_C_FLAGS_DEBUG "-O0 -g")
+
+IF("${ARCH}" STREQUAL "arm")
+ ADD_DEFINITIONS("-DTARGET")
+ENDIF("${ARCH}" STREQUAL "arm")
+
+ADD_DEFINITIONS("-DPREFIX=\"${CMAKE_INSTALL_PREFIX}\"")
+
+SET(CMAKE_EXE_LINKER_FLAGS "-Wl,--as-needed -Wl,--rpath=${LIB_INSTALL_DIR}")
+
+AUX_SOURCE_DIRECTORY (src MAIN_SRC)
+
+LIST (APPEND SOURCES
+ ${MAIN_SRC}
+)
+
+ADD_LIBRARY(${fw_name} SHARED ${SOURCES})
+
+TARGET_LINK_LIBRARIES(${fw_name} ${${fw_name}_LDFLAGS})
+
+SET_TARGET_PROPERTIES(${fw_name}
+ PROPERTIES
+ VERSION ${FULLVER}
+ SOVERSION ${MAJORVER}
+ CLEAN_DIRECT_OUTPUT 1
+)
+
+INSTALL(TARGETS ${fw_name} DESTINATION ${LIB_INSTALL_DIR})
+INSTALL(
+ DIRECTORY ${INC_DIR}/ DESTINATION include/media
+ FILES_MATCHING
+ PATTERN "webrtc_*.h" EXCLUDE
+ PATTERN "${INC_DIR}/*.h"
+)
+
+
+SET(PC_NAME ${fw_name})
+SET(PC_REQUIRED ${pc_dependents})
+SET(PC_LDFLAGS -l${fw_name})
+SET(PC_CFLAGS -I\${includedir}/media)
+
+CONFIGURE_FILE(
+ ${fw_name}.pc.in
+ ${CMAKE_CURRENT_SOURCE_DIR}/${fw_name}.pc
+ @ONLY
+)
+INSTALL(FILES ${CMAKE_CURRENT_SOURCE_DIR}/${fw_name}.pc DESTINATION ${LIB_INSTALL_DIR}/pkgconfig)
+
+ADD_SUBDIRECTORY(test)
+
+IF(UNIX)
+
+ADD_CUSTOM_TARGET (distclean @echo cleaning for source distribution)
+ADD_CUSTOM_COMMAND(
+ DEPENDS clean
+ COMMENT "distribution clean"
+ COMMAND find
+ ARGS .
+ -not -name config.cmake -and \(
+ -name tester.c -or
+ -name Testing -or
+ -name CMakeFiles -or
+ -name cmake.depends -or
+ -name cmake.check_depends -or
+ -name CMakeCache.txt -or
+ -name cmake.check_cache -or
+ -name *.cmake -or
+ -name Makefile -or
+ -name core -or
+ -name core.* -or
+ -name gmon.out -or
+ -name install_manifest.txt -or
+ -name *.pc -or
+ -name *~ \)
+ | grep -v TC | xargs rm -rf
+ TARGET distclean
+ VERBATIM
+)
+
+ENDIF(UNIX)
+
--- /dev/null
+Copyright (c) 2000 - 2020 Samsung Electronics Co., Ltd. All rights reserved.
+
+ Apache License
+ Version 2.0, January 2004
+ http://www.apache.org/licenses/
+
+ TERMS AND CONDITIONS FOR USE, REPRODUCTION, AND DISTRIBUTION
+
+ 1. Definitions.
+
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+ any Contribution intentionally submitted for inclusion in the Work
+ by You to the Licensor shall be under the terms and conditions of
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+ the terms of any separate license agreement you may have executed
+ with Licensor regarding such Contributions.
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+ 7. Disclaimer of Warranty. Unless required by applicable law or
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+ 8. Limitation of Liability. In no event and under no legal theory,
+ whether in tort (including negligence), contract, or otherwise,
+ unless required by applicable law (such as deliberate and grossly
+ negligent acts) or agreed to in writing, shall any Contributor be
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+ has been advised of the possibility of such damages.
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+ the Work or Derivative Works thereof, You may choose to offer,
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+ on Your own behalf and on Your sole responsibility, not on behalf
+ of any other Contributor, and only if You agree to indemnify,
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+
+ APPENDIX: How to apply the Apache License to your work.
+
+ To apply the Apache License to your work, attach the following
+ boilerplate notice, with the fields enclosed by brackets "[]"
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+ Licensed under the Apache License, Version 2.0 (the "License");
+ you may not use this file except in compliance with the License.
+ You may obtain a copy of the License at
+
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+
+ Unless required by applicable law or agreed to in writing, software
+ distributed under the License is distributed on an "AS IS" BASIS,
+ WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ See the License for the specific language governing permissions and
+ limitations under the License.
+
--- /dev/null
+Copyright (c) Samsung Electronics Co., Ltd. All rights reserved.
+Except as noted, this software is licensed under Apache License, Version 2.
+Please, see the LICENSE file for Apache License terms and conditions.
--- /dev/null
+# Package Information for pkg-config
+
+prefix=@PREFIX@
+exec_prefix=/usr
+libdir=@LIB_INSTALL_DIR@
+includedir=@INCLUDE_INSTALL_DIR@/media
+
+Name: @PC_NAME@
+Description: @PACKAGE_DESCRIPTION@
+Version: @VERSION@
+Requires: @PC_REQUIRED@
+Libs: -L${libdir} @PC_LDFLAGS@
+Cflags: -I${includedir}
+
--- /dev/null
+/*
+ * Copyright (c) 2020 Samsung Electronics Co., Ltd All Rights Reserved
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef __TIZEN_MEDIA_WEBRTC_H__
+#define __TIZEN_MEDIA_WEBRTC_H__
+
+#include <tizen.h>
+
+#ifdef __cplusplus
+extern "C" {
+#endif /* __cplusplus */
+
+#ifndef TIZEN_ERROR_MEDIA_WEBRTC
+#define TIZEN_ERROR_MEDIA_WEBRTC -0x03000000
+#endif
+
+/**
+* @file webrtc.h
+* @brief This file contains the WebRTC API.
+*/
+
+/**
+* @addtogroup CAPI_MEDIA_WEBRTC_MODULE
+* @{
+*/
+
+/**
+ * @brief WebRTC handle type.
+ * @since_tizen 6.0
+ */
+typedef void *webrtc_h;
+
+/**
+ * @brief Enumeration for WebRTC error.
+ * @since_tizen 6.0
+ */
+typedef enum
+{
+ WEBRTC_ERROR_NONE = TIZEN_ERROR_NONE, /**< Successful */
+ WEBRTC_ERROR_NOT_SUPPORTED = TIZEN_ERROR_NOT_SUPPORTED, /**< Not supported */
+ WEBRTC_ERROR_PERMISSION_DENIED = TIZEN_ERROR_PERMISSION_DENIED, /**< Permission denied */
+ WEBRTC_ERROR_INVALID_PARAMETER = TIZEN_ERROR_INVALID_PARAMETER, /**< Invalid parameter */
+ WEBRTC_ERROR_INVALID_OPERATION = TIZEN_ERROR_INVALID_OPERATION, /**< Invalid operation */
+ WEBRTC_ERROR_INVALID_STATE = TIZEN_ERROR_MEDIA_WEBRTC | 0x01, /**< Invalid state */
+ WEBRTC_ERROR_CONNECTION_FAILED = TIZEN_ERROR_MEDIA_WEBRTC | 0x02, /**< Connection failed */
+ WEBRTC_ERROR_RESOURCE_CONFLICT = TIZEN_ERROR_MEDIA_WEBRTC | 0x03, /**< Resource conflict */
+} webrtc_error_e;
+
+/**
+ * @brief Enumeration for WebRTC state.
+ * @since_tizen 6.0
+ */
+typedef enum
+{
+ WEBRTC_STATE_IDLE, /**< Created but not started */
+ WEBRTC_STATE_PLAYING, /**< Started */
+} webrtc_state_e;
+
+/**
+ * @brief Enumeration for WebRTC media source type.
+ * @since_tizen 6.0
+ */
+typedef enum {
+ WEBRTC_MEDIA_SOURCE_TYPE_CAMERA,
+ WEBRTC_MEDIA_SOURCE_TYPE_MIC,
+ WEBRTC_MEDIA_SOURCE_TYPE_AUDIOTEST,
+ WEBRTC_MEDIA_SOURCE_TYPE_VIDEOTEST
+} webrtc_media_source_type_e;
+
+/**
+ * @brief Creates an instance of WebRTC.
+ * @since_tizen 6.0
+ * @remarks You must release @a webrtc using webrtc_destroy().
+ * @param [out] webrtc WebRTC handle
+ * @return @c 0 on success,
+ * otherwise a negative error value
+ * @retval #WEBRTC_ERROR_NONE Successful
+ * @retval #WEBRTC_ERROR_INVALID_PARAMETER Invalid parameter
+ * @retval #WEBRTC_ERROR_INVALID_OPERATION Invalid operation
+ * @post @a webrtc state will be #WEBRTC_STATE_IDLE.
+ * @see webrtc_destroy()
+ */
+int webrtc_create(webrtc_h *webrtc);
+
+/**
+ * @brief Starts the WebRTC.
+ * @since_tizen 6.0
+ * @param [in] webrtc WebRTC handle
+ * @return @c 0 on success,
+ * otherwise a negative error value
+ * @retval #WEBRTC_ERROR_NONE Successful
+ * @retval #WEBRTC_ERROR_INVALID_PARAMETER Invalid parameter
+ * @retval #WEBRTC_ERROR_INVALID_OPERATION Invalid operation
+ * @retval #WEBRTC_ERROR_INVALID_STATE Invalid state
+ * @pre @a webrtc state must be set to #WEBRTC_STATE_IDLE.
+ * @post @a webrtc state will be #WEBRTC_STATE_PLAYING.
+ * @see webrtc_create()
+ * @see webrtc_stop()
+ */
+int webrtc_start(webrtc_h webrtc);
+
+/**
+ * @brief Stops the WebRTC.
+ * @since_tizen 6.0
+ * @param [in] webrtc WebRTC handle
+ * @return @c 0 on success,
+ * otherwise a negative error value
+ * @retval #WEBRTC_ERROR_NONE Successful
+ * @retval #WEBRTC_ERROR_INVALID_PARAMETER Invalid parameter
+ * @retval #WEBRTC_ERROR_INVALID_OPERATION Invalid operation
+ * @retval #WEBRTC_ERROR_INVALID_STATE Invalid state
+ * @pre @a webrtc state will be #WEBRTC_STATE_PLAYING.
+ * @post @a webrtc state will be #WEBRTC_STATE_IDLE.
+ * @see webrtc_create()
+ * @see webrtc_start()
+ */
+int webrtc_stop(webrtc_h webrtc);
+
+/**
+ * @brief Destroys the WebRTC.
+ * @since_tizen 6.0
+ * @param [in] webrtc WebRTC handle
+ * @return @c 0 on success,
+ * otherwise a negative error value
+ * @retval #WEBRTC_ERROR_NONE Successful
+ * @retval #WEBRTC_ERROR_INVALID_PARAMETER Invalid parameter
+ * @retval #WEBRTC_ERROR_INVALID_OPERATION Invalid operation
+ * @retval #WEBRTC_ERROR_INVALID_STATE Invalid state
+ * @see webrtc_create()
+ */
+int webrtc_destroy(webrtc_h webrtc);
+
+/**
+ * @brief Gets the WebRTC state.
+ * @since_tizen 6.0
+ * @param [in] webrtc WebRTC handle
+ * @param [out] state WebRTC state
+ * @return @c 0 on success,
+ * otherwise a negative error value
+ * @retval #WEBRTC_ERROR_NONE Successful
+ * @retval #WEBRTC_ERROR_INVALID_PARAMETER Invalid parameter
+ * @retval #WEBRTC_ERROR_INVALID_OPERATION Invalid operation
+ * @see #webrtc_state_e
+ */
+int webrtc_get_state(webrtc_h webrtc, webrtc_state_e *state);
+
+/**
+ * @brief Adds a media source.
+ * @since_tizen 6.0
+ * @param [in] webrtc WebRTC handle
+ * @param [in] type The media source type to be added
+ * @param [out] source_id The media source id
+ * @return @c 0 on success,
+ * otherwise a negative error value
+ * @retval #WEBRTC_ERROR_NONE Successful
+ * @retval #WEBRTC_ERROR_INVALID_PARAMETER Invalid parameter
+ * @retval #WEBRTC_ERROR_INVALID_OPERATION Invalid operation
+ * @retval #WEBRTC_ERROR_INVALID_STATE Invalid state
+ * @pre @a webrtc state must be set to #WEBRTC_STATE_IDLE.
+ * @see webrtc_remove_media_source()
+ */
+int webrtc_add_media_source(webrtc_h webrtc, webrtc_media_source_type_e type, int *source_id);
+
+/**
+ * @brief Removes the media source.
+ * @since_tizen 6.0
+ * @param [in] webrtc WebRTC handle
+ * @param [in] source_id The media source id to be removed
+ * @return @c 0 on success,
+ * otherwise a negative error value
+ * @retval #WEBRTC_ERROR_NONE Successful
+ * @retval #WEBRTC_ERROR_INVALID_PARAMETER Invalid parameter
+ * @retval #WEBRTC_ERROR_INVALID_OPERATION Invalid operation
+ * @retval #WEBRTC_ERROR_INVALID_STATE Invalid state
+ * @pre @a webrtc state must be set to #WEBRTC_STATE_IDLE.
+ * @pre Add media source to @a webrtc by calling webrtc_add_media_source().
+ * @see webrtc_add_media_source()
+ */
+int webrtc_remove_media_source(webrtc_h webrtc, int source_id);
+
+/**
+ * @brief Sets a STUN server URL.
+ * @since_tizen 6.0
+ * @param [in] webrtc WebRTC handle
+ * @param [in] stun_server The STUN server URL
+ * @return @c 0 on success,
+ * otherwise a negative error value
+ * @retval #WEBRTC_ERROR_NONE Successful
+ * @retval #WEBRTC_ERROR_INVALID_PARAMETER Invalid parameter
+ * @retval #WEBRTC_ERROR_INVALID_STATE Invalid state
+ * @pre @a webrtc state must be set to #WEBRTC_STATE_IDLE.
+ */
+int webrtc_set_stun_server(webrtc_h webrtc, const char *stun_server);
+
+/**
+ * @}
+ */
+
+#ifdef __cplusplus
+}
+#endif /* __cplusplus */
+
+#endif /* __TIZEN_MEDIA_WEBRTC_H__ */
--- /dev/null
+/*
+ * Copyright (c) 2020 Samsung Electronics Co., Ltd All Rights Reserved
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef __TIZEN_MEDIA_WEBRTC_PRIVATE_H__
+#define __TIZEN_MEDIA_WEBRTC_PRIVATE_H__
+
+#include <gst/gst.h>
+#include <stdio.h>
+#include <dlog.h>
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+#ifdef LOG_TAG
+#undef LOG_TAG
+#endif
+
+#define LOG_TAG "TIZEN_N_WEBRTC"
+
+#define FONT_COLOR_RESET "\033[0m"
+#define FONT_COLOR_RED "\033[31m"
+#define FONT_COLOR_GREEN "\033[32m"
+#define FONT_COLOR_YELLOW "\033[33m"
+#define FONT_COLOR_BLUE "\033[34m"
+#define FONT_COLOR_PURPLE "\033[35m"
+#define FONT_COLOR_CYAN "\033[36m"
+#define FONT_COLOR_GRAY "\033[37m"
+
+#define LOG_DEBUG(fmt, arg...) \
+do { \
+ LOGD(FONT_COLOR_RESET""fmt""FONT_COLOR_RESET, ##arg); \
+} while (0)
+
+#define LOG_INFO(fmt, arg...) \
+do { \
+ LOGI(FONT_COLOR_GREEN""fmt""FONT_COLOR_RESET, ##arg); \
+} while (0)
+
+#define LOG_WARNING(fmt, arg...) \
+do { \
+ LOGW(FONT_COLOR_YELLOW""fmt""FONT_COLOR_RESET, ##arg); \
+} while (0)
+
+#define LOG_ERROR(fmt, arg...) \
+do { \
+ LOGE(FONT_COLOR_RED""fmt""FONT_COLOR_RESET, ##arg); \
+} while (0)
+
+#define LOG_DEBUG_ENTER() \
+do { \
+ LOGD(FONT_COLOR_PURPLE"<Enter>"FONT_COLOR_RESET); \
+} while (0)
+
+#define LOG_DEBUG_LEAVE() \
+do { \
+ LOGD(FONT_COLOR_PURPLE"<Leave>"FONT_COLOR_RESET); \
+} while (0)
+
+#define RET_IF(expr, fmt, arg...) \
+do { \
+ if ((expr)) { \
+ LOG_ERROR(""fmt"", ##arg); \
+ return; \
+ } \
+} while (0)
+
+#define RET_VAL_IF(expr, val, fmt, arg...) \
+do { \
+ if ((expr)) { \
+ LOG_ERROR(""fmt"", ##arg); \
+ return (val);\
+ } \
+} while (0)
+
+#define RET_VAL_WITH_UNLOCK_IF(expr, val, mutex, fmt, arg...) \
+do { \
+ if ((expr)) { \
+ LOG_ERROR(""fmt"", ##arg); \
+ g_mutex_unlock(mutex); \
+ return (val);\
+ } \
+} while (0)
+
+#define SAFE_FREE(src) { if (src) { free(src); src = NULL; } }
+
+#define SOURCES_MAX 8
+
+typedef struct _webrtc_ini_s {
+ gboolean generate_dot;
+ gchar **gst_args;
+} webrtc_ini_s;
+
+typedef struct _webrtc_gst_src_s {
+ int source_id;
+ GstElement *bin;
+} webrtc_gst_src_s;
+
+typedef struct _webrtc_gst_s {
+ GstElement *pipeline;
+ GstBus *bus;
+ guint bus_watcher;
+ webrtc_gst_src_s *audiosrc[SOURCES_MAX];
+ webrtc_gst_src_s *videosrc[SOURCES_MAX];
+} webrtc_gst_s;
+
+typedef struct _webrtc_s {
+ webrtc_ini_s ini;
+
+ GMutex mutex;
+
+ webrtc_gst_s gst;
+
+ webrtc_state_e state;
+ webrtc_state_e pend_state;
+} webrtc_s;
+
+int _ini_load(webrtc_s *webrtc);
+int _gst_init(webrtc_s *webrtc);
+int _gst_build_pipeline(webrtc_s *webrtc);
+
+#ifdef __cplusplus
+}
+#endif /* __cplusplus */
+
+#endif /* __TIZEN_MEDIA_WEBRTC_PRIVATE_H__ */
--- /dev/null
+<manifest>
+ <request>
+ <domain name="_" />
+ </request>
+ <assign>
+ <filesystem path="/usr/bin/webrtc_test" label="_" exec_label="none" />
+ </assign>
+</manifest>
--- /dev/null
+Name: capi-media-webrtc
+Summary: A WebRTC library in Tizen Native API
+Version: 0.1.1
+Release: 0
+Group: Multimedia/API
+License: Apache-2.0
+URL: http://source.tizen.org
+Source0: %{name}-%{version}.tar.gz
+Source1001: capi-media-webrtc.manifest
+BuildRequires: cmake
+BuildRequires: pkgconfig(dlog)
+BuildRequires: pkgconfig(glib-2.0)
+BuildRequires: pkgconfig(capi-base-common)
+BuildRequires: pkgconfig(gstreamer-1.0)
+BuildRequires: pkgconfig(appcore-efl)
+BuildRequires: pkgconfig(elementary)
+
+%description
+A WebRTC library in Tizen Native API.
+
+%package devel
+Summary: Multimedia WebRTC Library in Tizen Native API (Development)
+Group: TO_BE/FILLED_IN
+Requires: %{name} = %{version}-%{release}
+
+%description devel
+WebRTC Library in Tizen Native API (DEV).
+
+%prep
+%setup -q
+cp %{SOURCE1001} .
+
+%build
+MAJORVER=`echo %{version} | awk 'BEGIN {FS="."}{print $1}'`
+cmake . -DCMAKE_INSTALL_PREFIX=/usr -DFULLVER=%{version} -DMAJORVER=${MAJORVER}
+
+make %{?jobs:-j%jobs}
+
+%install
+rm -rf %{buildroot}
+mkdir -p %{buildroot}/usr/share/license
+mkdir -p %{buildroot}/usr/bin
+cp LICENSE.APLv2 %{buildroot}/usr/share/license/%{name}
+cp test/webrtc_test %{buildroot}/usr/bin
+
+%make_install
+
+%post
+/sbin/ldconfig
+
+%postun -p /sbin/ldconfig
+
+
+%files
+%manifest capi-media-webrtc.manifest
+%{_libdir}/libcapi-media-webrtc.so.*
+%{_datadir}/license/%{name}
+/usr/bin/*
+
+%files devel
+%{_includedir}/media/*.h
+%{_libdir}/pkgconfig/*.pc
+%{_libdir}/libcapi-media-webrtc.so
+
+
--- /dev/null
+/*
+ * Copyright (c) 2020 Samsung Electronics Co., Ltd All Rights Reserved
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#include <stdlib.h>
+#include <string.h>
+#include <dlog.h>
+
+#include "webrtc.h"
+#include "webrtc_private.h"
+
+int webrtc_create(webrtc_h *webrtc)
+{
+ int ret = WEBRTC_ERROR_NONE;
+ webrtc_s *_webrtc = NULL;
+
+ RET_VAL_IF(webrtc == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "webrtc is NULL");
+
+ _webrtc = g_new0(webrtc_s, 1);
+
+ g_mutex_init(&_webrtc->mutex);
+ g_mutex_lock(&_webrtc->mutex);
+
+ _ini_load(_webrtc);
+ _gst_init(_webrtc);
+ _gst_build_pipeline(_webrtc);
+
+ _webrtc->state = WEBRTC_STATE_IDLE;
+
+ *webrtc = _webrtc;
+
+ g_mutex_unlock(&_webrtc->mutex);
+
+ LOG_INFO("webrtc[%p] is created", *webrtc);
+
+ return ret;
+}
+
+int webrtc_destroy(webrtc_h webrtc)
+{
+ webrtc_s *_webrtc = (webrtc_s*)webrtc;
+
+ RET_VAL_IF(_webrtc == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "webrtc is NULL");
+
+ g_mutex_lock(&_webrtc->mutex);
+
+ /* set state to null here */
+
+ if (_webrtc->gst.bus)
+ gst_object_unref(_webrtc->gst.bus);
+ if (_webrtc->gst.pipeline)
+ gst_object_unref(_webrtc->gst.pipeline);
+
+ g_mutex_unlock(&_webrtc->mutex);
+ g_mutex_clear(&_webrtc->mutex);
+
+ g_free(_webrtc);
+
+ LOG_INFO("webrtc[%p] is destroyed", webrtc);
+
+ return WEBRTC_ERROR_NONE;
+}
+
+int webrtc_start(webrtc_h webrtc)
+{
+ webrtc_s *_webrtc = (webrtc_s*)webrtc;
+
+ RET_VAL_IF(_webrtc == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "webrtc is NULL");
+
+ g_mutex_lock(&_webrtc->mutex);
+
+ RET_VAL_WITH_UNLOCK_IF(_webrtc->state != WEBRTC_STATE_IDLE, WEBRTC_ERROR_INVALID_STATE, &_webrtc->mutex, "the state should be IDLE");
+
+ /* Implementation */
+
+ g_mutex_unlock(&_webrtc->mutex);
+
+ LOG_INFO("webrtc[%p] is started", webrtc);
+
+ return WEBRTC_ERROR_NONE;
+}
+
+int webrtc_stop(webrtc_h webrtc)
+{
+ webrtc_s *_webrtc = (webrtc_s*)webrtc;
+
+ RET_VAL_IF(_webrtc == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "webrtc is NULL");
+
+ LOG_DEBUG_ENTER();
+
+ g_mutex_lock(&_webrtc->mutex);
+
+ RET_VAL_WITH_UNLOCK_IF(_webrtc->state != WEBRTC_STATE_PLAYING, WEBRTC_ERROR_INVALID_STATE, &_webrtc->mutex, "the state should be PLAYING");
+
+ /* Implementation */
+
+ g_mutex_unlock(&_webrtc->mutex);
+
+ LOG_INFO("webrtc[%p] is stopped", webrtc);
+
+ return WEBRTC_ERROR_NONE;
+}
+
+int webrtc_get_state(webrtc_h webrtc, webrtc_state_e *state)
+{
+ webrtc_s *_webrtc = (webrtc_s*)webrtc;
+
+ RET_VAL_IF(_webrtc == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "webrtc is NULL");
+
+ *state = _webrtc->state;
+
+ LOG_INFO("state[%d]", *state);
+
+ return WEBRTC_ERROR_NONE;
+}
+
+int webrtc_add_media_source(webrtc_h webrtc, webrtc_media_source_type_e type, int *source_id)
+{
+ webrtc_s *_webrtc = (webrtc_s*)webrtc;
+
+ RET_VAL_IF(_webrtc == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "webrtc is NULL");
+ RET_VAL_IF(source_id == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "source_id is NULL");
+
+ g_mutex_lock(&_webrtc->mutex);
+
+ RET_VAL_WITH_UNLOCK_IF(_webrtc->state != WEBRTC_STATE_IDLE, WEBRTC_ERROR_INVALID_STATE, &_webrtc->mutex, "the state should be IDLE");
+
+ /* Implementation */
+
+ g_mutex_unlock(&_webrtc->mutex);
+
+ LOG_INFO("source_id[%d]", *source_id);
+
+ return WEBRTC_ERROR_NONE;
+}
+
+int webrtc_remove_media_source(webrtc_h webrtc, int source_id)
+{
+ webrtc_s *_webrtc = (webrtc_s*)webrtc;
+
+ RET_VAL_IF(_webrtc == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "webrtc is NULL");
+
+ g_mutex_lock(&_webrtc->mutex);
+
+ RET_VAL_WITH_UNLOCK_IF(_webrtc->state != WEBRTC_STATE_IDLE, WEBRTC_ERROR_INVALID_STATE, &_webrtc->mutex, "the state should be IDLE");
+
+ /* Implementation */
+
+ g_mutex_unlock(&_webrtc->mutex);
+
+ LOG_INFO("source_id[%d]", source_id);
+
+ return WEBRTC_ERROR_NONE;
+}
+
+int webrtc_set_stun_server(webrtc_h webrtc, const char *stun_server)
+{
+ webrtc_s *_webrtc = (webrtc_s*)webrtc;
+
+ RET_VAL_IF(_webrtc == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "webrtc is NULL");
+ RET_VAL_IF(stun_server == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "stun_server is NULL");
+
+ g_mutex_lock(&_webrtc->mutex);
+
+ RET_VAL_WITH_UNLOCK_IF(_webrtc->state != WEBRTC_STATE_IDLE, WEBRTC_ERROR_INVALID_STATE, &_webrtc->mutex, "the state should be IDLE");
+
+ /* Implementation */
+
+ g_mutex_unlock(&_webrtc->mutex);
+
+ LOG_INFO("stun_server[%s]", stun_server);
+
+ return WEBRTC_ERROR_NONE;
+}
\ No newline at end of file
--- /dev/null
+/*
+ * Copyright (c) 2020 Samsung Electronics Co., Ltd All Rights Reserved
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#include "webrtc.h"
+#include "webrtc_private.h"
+
+static gboolean __bus_watch_cb(GstBus *bus, GstMessage *message, gpointer user_data)
+{
+ webrtc_s *webrtc = (webrtc_s *)user_data;
+ GError *err = NULL;
+ GstState gst_state_old = GST_STATE_VOID_PENDING;
+ GstState gst_state_new = GST_STATE_VOID_PENDING;
+ GstState gst_state_pending = GST_STATE_VOID_PENDING;
+ gchar *state_transition_name = NULL;
+
+ RET_VAL_IF(webrtc == NULL, FALSE, "webrtc is NULL");
+ RET_VAL_IF(webrtc->gst.pipeline == NULL, FALSE, "pipeline is NULL");
+
+ if (message == NULL) {
+ LOG_DEBUG("message is null");
+ return TRUE;
+ }
+
+ switch (GST_MESSAGE_TYPE(message)) {
+ case GST_MESSAGE_ERROR:
+ gst_message_parse_error(message, &err, NULL);
+
+ LOG_ERROR("Error from [%s]: %s", GST_OBJECT_NAME(GST_OBJECT_CAST(GST_ELEMENT(GST_MESSAGE_SRC(message)))), err->message);
+ /* FIXME : invoke error cb */
+
+ g_error_free(err);
+ break;
+
+ case GST_MESSAGE_STATE_CHANGED:
+ if (GST_MESSAGE_SRC(message) != GST_OBJECT(webrtc->gst.pipeline))
+ return TRUE;
+
+ gst_message_parse_state_changed(message, &gst_state_old, &gst_state_new, &gst_state_pending);
+ state_transition_name = g_strdup_printf("Old[GST_STATE_%s] New[GST_STATE_%s] Pending[GST_STATE_%s]",
+ gst_element_state_get_name(gst_state_old), gst_element_state_get_name(gst_state_new),
+ gst_element_state_get_name(gst_state_pending));
+
+ LOG_INFO("GST_MESSAGE_STATE_CHANGED: %s", state_transition_name);
+
+ /* FIXME : set pipeline state */
+
+ g_free(state_transition_name);
+ break;
+
+ case GST_MESSAGE_ASYNC_DONE:
+ if (GST_MESSAGE_SRC(message) != GST_OBJECT(webrtc->gst.pipeline))
+ return TRUE;
+
+ LOG_INFO("GST_MESSAGE_ASYNC_DONE");
+
+ break;
+
+ case GST_MESSAGE_EOS:
+ LOG_INFO("GST_MESSAGE_EOS end-of-stream");
+
+ break;
+
+ default:
+ break;
+ }
+
+ return TRUE;
+}
+
+int _ini_load(webrtc_s *webrtc)
+{
+ RET_VAL_IF(webrtc == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "webrtc is NULL");
+
+ memset(&webrtc->ini, 0, sizeof(webrtc_ini_s));
+
+ /* FIXME: load from ini file */
+
+ webrtc->ini.generate_dot = FALSE;
+
+ return WEBRTC_ERROR_NONE;
+}
+
+int _gst_init(webrtc_s *webrtc)
+{
+ int i = 0;
+ int argc = 1;
+ char **argv = NULL;
+ GError *err = NULL;
+ gboolean gst_ret = 0;
+
+ RET_VAL_IF(webrtc == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "webrtc is NULL");
+
+ /* get argc(number of command line option), argc is always one without option */
+ if (webrtc->ini.gst_args)
+ argc += g_strv_length(webrtc->ini.gst_args);
+
+ argv = (char **)calloc(argc, sizeof(char*));
+ RET_VAL_IF(argv == NULL, WEBRTC_ERROR_INVALID_OPERATION, "failed to calloc()");
+
+ argv[0] = g_strdup("capi-media-webrtc");
+
+ if (webrtc->ini.gst_args) {
+ for ( ; webrtc->ini.gst_args[i]; ++i) {
+ if (argc <= i+1) {
+ LOG_ERROR("need to check, prevent overrun");
+ break;
+ }
+ argv[i+1] = webrtc->ini.gst_args[i];
+ LOG_DEBUG("add [%s] gstreamer parameter", argv[i+1]);
+ }
+ }
+
+ gst_ret = gst_init_check(&argc, &argv, &err);
+
+ for (i = 1; i < argc; i++)
+ argv[i] = NULL;
+
+ SAFE_FREE(argv[0]);
+ SAFE_FREE(argv);
+
+ if (!gst_ret) {
+ LOG_ERROR("failed to gst_init_check(), err[%s]", err->message);
+ g_clear_error(&err);
+ return WEBRTC_ERROR_INVALID_OPERATION;
+ }
+
+ return WEBRTC_ERROR_NONE;
+}
+
+int _gst_build_pipeline(webrtc_s *webrtc)
+{
+ int ret = WEBRTC_ERROR_NONE;
+
+ RET_VAL_IF(webrtc == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "webrtc is NULL");
+
+ webrtc->gst.pipeline = gst_pipeline_new("native-webrtc-pipeline");
+ RET_VAL_IF(webrtc->gst.pipeline == NULL, WEBRTC_ERROR_INVALID_OPERATION, "pipeline is NULL");
+
+ webrtc->gst.bus = gst_pipeline_get_bus(GST_PIPELINE(webrtc->gst.pipeline));
+ RET_VAL_IF(webrtc->gst.bus == NULL, WEBRTC_ERROR_INVALID_OPERATION, "bus is NULL");
+
+ webrtc->gst.bus_watcher = gst_bus_add_watch(webrtc->gst.bus, (GstBusFunc) __bus_watch_cb, webrtc);
+
+ /* FIXME : initial bin setting */
+
+ return ret;
+}
--- /dev/null
+CMAKE_MINIMUM_REQUIRED(VERSION 2.6)
+SET(fw_test "${fw_name}-test")
+
+INCLUDE_DIRECTORIES(../include)
+
+link_directories(${CMAKE_SOURCE_DIR}/../)
+
+INCLUDE(FindPkgConfig)
+pkg_check_modules(${fw_test} REQUIRED glib-2.0 appcore-efl elementary)
+FOREACH(flag ${${fw_test}_CFLAGS})
+ SET(EXTRA_CFLAGS "${EXTRA_CFLAGS} ${flag}")
+ENDFOREACH(flag)
+
+SET(CMAKE_C_FLAGS "${CMAKE_C_FLAGS} ${EXTRA_CFLAGS} -pie")
+
+aux_source_directory(. sources)
+
+FOREACH(src ${sources})
+ GET_FILENAME_COMPONENT(src_name ${src} NAME_WE)
+ MESSAGE("${src_name}")
+ ADD_EXECUTABLE(${src_name} ${src})
+ TARGET_LINK_LIBRARIES(${src_name} capi-media-webrtc ${${fw_test}_LDFLAGS})
+ENDFOREACH()
+
--- /dev/null
+/*
+* Copyright (c) 2020 Samsung Electronics Co., Ltd All Rights Reserved
+*
+* Licensed under the Apache License, Version 2.0 (the "License");
+* you may not use this file except in compliance with the License.
+* You may obtain a copy of the License at
+*
+* http://www.apache.org/licenses/LICENSE-2.0
+*
+* Unless required by applicable law or agreed to in writing, software
+* distributed under the License is distributed on an "AS IS" BASIS,
+* WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+* See the License for the specific language governing permissions and
+* limitations under the License.
+*/
+#define USE_EVENT_HANDLER
+
+#include <webrtc.h>
+#include <appcore-efl.h>
+#include <Elementary.h>
+#include <glib.h>
+
+#ifdef PACKAGE
+#undef PACKAGE
+#endif
+#define PACKAGE "webrtc_test"
+
+#ifdef LOG_TAG
+#undef LOG_TAG
+#endif
+#define LOG_TAG "WEBRTC_TEST"
+
+#define MAX_STRING_LEN 512
+
+enum {
+ CURRENT_STATUS_MAINMENU,
+ CURRENT_STATUS_ADD_MEDIA_SOURCE,
+ CURRENT_STATUS_REMOVE_MEDIA_SOURCE,
+ CURRENT_STATUS_SET_STUN_SERVER,
+ CURRENT_STATUS_SETTING_SIGNALLING_SERVER,
+ CURRENT_STATUS_SETTING_PROXY,
+};
+
+/* for video display */
+static Evas_Object *g_eo;
+static Evas_Object *g_win_id;
+static Evas_Object *g_selected_win_id;
+
+typedef struct {
+ Evas_Object *win;
+ Evas_Object *layout_main;
+} appdata;
+
+static appdata ad;
+static webrtc_h g_webrtc;
+static gchar g_signalling_server[MAX_STRING_LEN];
+static gchar g_stun_server[MAX_STRING_LEN];
+static gchar g_proxy[MAX_STRING_LEN];
+int g_menu_state = CURRENT_STATUS_MAINMENU;
+
+static void win_del(void *data, Evas_Object *obj, void *event)
+{
+ elm_exit();
+}
+
+static Evas_Object *create_win(const char *name)
+{
+ Evas_Object *eo = NULL;
+ int w = 0;
+ int h = 0;
+
+ g_print("[%s][%d] name=%s\n", __func__, __LINE__, name);
+
+ eo = elm_win_add(NULL, name, ELM_WIN_BASIC);
+ if (eo) {
+ elm_win_title_set(eo, name);
+ elm_win_borderless_set(eo, EINA_TRUE);
+ evas_object_smart_callback_add(eo, "delete,request", win_del, NULL);
+ elm_win_screen_size_get(eo, NULL, NULL, &w, &h);
+ g_print("window size :%d,%d", w, h);
+ evas_object_resize(eo, w, h);
+ elm_win_autodel_set(eo, EINA_TRUE);
+ elm_win_alpha_set(eo, EINA_TRUE);
+ }
+ return eo;
+}
+
+static Evas_Object *create_image_object(Evas_Object *eo_parent)
+{
+ if (!eo_parent)
+ return NULL;
+
+ Evas *evas = evas_object_evas_get(eo_parent);
+ Evas_Object *eo = NULL;
+
+ eo = evas_object_image_add(evas);
+
+ return eo;
+}
+
+void create_render_rect_and_bg(Evas_Object *win)
+{
+ if (!win) {
+ g_print("no win");
+ return;
+ }
+ Evas_Object *bg, *rect;
+
+ bg = elm_bg_add(win);
+ elm_win_resize_object_add(win, bg);
+ evas_object_size_hint_weight_set(bg, EVAS_HINT_EXPAND, EVAS_HINT_EXPAND);
+ evas_object_show(bg);
+
+ rect = evas_object_rectangle_add(evas_object_evas_get(win));
+ if (!rect) {
+ g_print("no rect");
+ return;
+ }
+ evas_object_color_set(rect, 0, 0, 0, 0);
+ evas_object_render_op_set(rect, EVAS_RENDER_COPY);
+
+ elm_win_resize_object_add(win, rect);
+ evas_object_size_hint_weight_set(rect, EVAS_HINT_EXPAND, EVAS_HINT_EXPAND);
+ evas_object_show(rect);
+ evas_object_show(win);
+}
+
+static int app_create(void *data)
+{
+ appdata *ad = data;
+ Evas_Object *win = NULL;
+
+ /* use gl backend */
+ elm_config_accel_preference_set("opengl");
+
+ /* create window */
+ win = create_win(PACKAGE);
+ if (win == NULL)
+ return -1;
+ ad->win = win;
+ g_win_id = win;
+ g_selected_win_id = g_win_id;
+ create_render_rect_and_bg(ad->win);
+ /* Create evas image object for EVAS surface */
+ g_eo = create_image_object(ad->win);
+ evas_object_image_size_set(g_eo, 500, 500);
+ evas_object_image_fill_set(g_eo, 0, 0, 500, 500);
+ evas_object_resize(g_eo, 500, 500);
+
+ elm_win_activate(win);
+ evas_object_show(win);
+
+ return 0;
+}
+
+static int app_terminate(void *data)
+{
+ appdata *ad = data;
+
+ if (g_win_id) {
+ evas_object_del(g_win_id);
+ g_win_id = NULL;
+ }
+
+ ad->win = NULL;
+ g_selected_win_id = NULL;
+
+ return 0;
+}
+
+struct appcore_ops ops = {
+ .create = app_create,
+ .terminate = app_terminate,
+};
+
+static void _webrtc_create()
+{
+ int ret = WEBRTC_ERROR_NONE;
+
+ if (g_webrtc) {
+ g_print("failed to _webrtc_create(), already created\n");
+ return;
+ }
+
+ ret = webrtc_create(&g_webrtc);
+ if (ret != WEBRTC_ERROR_NONE)
+ g_print("webrtc_create() returned [0x%x]\n", ret);
+ else
+ g_print("webrtc_create() success, g_webrtc[%p]\n", g_webrtc);
+}
+
+static void _webrtc_destroy()
+{
+ int ret = WEBRTC_ERROR_NONE;
+
+ if (!g_webrtc) {
+ g_print("failed to _webrtc_destroy(), g_webrtc is NULL\n");
+ return;
+ }
+
+ ret = webrtc_destroy(g_webrtc);
+ if (ret != WEBRTC_ERROR_NONE) {
+ g_print("webrtc_destroy() returned [0x%x]\n", ret);
+ } else {
+ g_print("webrtc_destroy() success\n");
+ g_webrtc = NULL;
+ }
+}
+
+static void _webrtc_start()
+{
+ int ret = WEBRTC_ERROR_NONE;
+
+ if (!g_webrtc) {
+ g_print("failed to _webrtc_start(), g_webrtc is NULL\n");
+ return;
+ }
+
+ ret = webrtc_start(g_webrtc);
+ if (ret != WEBRTC_ERROR_NONE)
+ g_print("webrtc_start() returned [0x%x]\n", ret);
+ else
+ g_print("webrtc_start() success\n");
+}
+
+static void _webrtc_stop()
+{
+ int ret = WEBRTC_ERROR_NONE;
+
+ if (!g_webrtc) {
+ g_print("failed to _webrtc_stop(), g_webrtc is NULL\n");
+ return;
+ }
+
+ ret = webrtc_stop(g_webrtc);
+ if (ret != WEBRTC_ERROR_NONE)
+ g_print("webrtc_stop() returned [0x%x]\n", ret);
+ else
+ g_print("webrtc_stop() success\n");
+}
+
+static void _webrtc_get_state()
+{
+ int ret = WEBRTC_ERROR_NONE;
+ webrtc_state_e state;
+
+ ret = webrtc_get_state(g_webrtc, &state);
+ if (ret != WEBRTC_ERROR_NONE)
+ g_print("webrtc_get_state() returned [0x%x]\n", ret);
+ else
+ g_print("state: [%d]\n", state);
+}
+
+static void _webrtc_add_media_source(int type)
+{
+ int ret = WEBRTC_ERROR_NONE;
+ int source_id = 0;
+
+ ret = webrtc_add_media_source(g_webrtc, (webrtc_media_source_type_e)type, &source_id);
+ if (ret != WEBRTC_ERROR_NONE)
+ g_print("failed to webrtc_add_media_source(), ret[0x%x]\n", ret);
+ else
+ g_print("webrtc_add_media_source() success, source_id[%d]\n", source_id);
+}
+
+static void _webrtc_remove_media_source(int source_id)
+{
+ int ret = WEBRTC_ERROR_NONE;
+
+ ret = webrtc_remove_media_source(g_webrtc, source_id);
+ if (ret != WEBRTC_ERROR_NONE)
+ g_print("failed to webrtc_remove_media_source(), source_id[%d], ret[0x%x]\n", source_id, ret);
+ else
+ g_print("webrtc_remove_media_source() success, source_id[%d]\n", source_id);
+}
+
+static int __copy_string_arr(gchar *dest_arr, char *string)
+{
+ int len = 0;
+ gsize src_size = 0;
+
+ if (!string)
+ return -1;
+
+ len = strlen(string);
+
+ src_size = g_strlcpy(dest_arr, string, MAX_STRING_LEN);
+ if (src_size != len) {
+ g_print("failed to g_strlcpy()\n");
+ return -1;
+ }
+
+ return 0;
+}
+
+static void _webrtc_set_stun_server(char *uri)
+{
+ int ret = 0;
+
+ if (!uri)
+ return;
+
+ ret = __copy_string_arr(g_stun_server, uri);
+ if (ret != 0)
+ return;
+
+ ret = webrtc_set_stun_server(g_webrtc, g_stun_server);
+ if (ret != WEBRTC_ERROR_NONE)
+ g_print("failed to webrtc_set_stun_server(), uri[%s]\n", g_stun_server);
+ else
+ g_print("webrtc_set_stun_server() success, uri[%s]\n", g_stun_server);
+}
+
+static void _setting_signalling_server(char *uri)
+{
+ int ret = 0;
+
+ if (!uri)
+ return;
+
+ ret = __copy_string_arr(g_signalling_server, uri);
+ if (ret != 0)
+ return;
+}
+
+static void _setting_proxy(char *uri)
+{
+ int ret = 0;
+
+ if (!uri)
+ return;
+
+ ret = __copy_string_arr(g_proxy, uri);
+ if (ret != 0)
+ return;
+}
+
+void quit_program()
+{
+ if (g_webrtc != NULL) {
+ _webrtc_stop();
+ _webrtc_destroy();
+ }
+
+ elm_exit();
+}
+
+void _interpret_main_menu(char *cmd)
+{
+ int len = strlen(cmd);
+ if (len == 1) {
+ if (strncmp(cmd, "c", 1) == 0) {
+ _webrtc_create();
+
+ } else if (strncmp(cmd, "a", 1) == 0) {
+ g_menu_state = CURRENT_STATUS_ADD_MEDIA_SOURCE;
+
+ } else if (strncmp(cmd, "r", 1) == 0) {
+ g_menu_state = CURRENT_STATUS_REMOVE_MEDIA_SOURCE;
+
+ } else if (strncmp(cmd, "s", 1) == 0) {
+ _webrtc_start();
+
+ } else if (strncmp(cmd, "t", 1) == 0) {
+ _webrtc_stop();
+
+ } else if (strncmp(cmd, "d", 1) == 0) {
+ _webrtc_destroy();
+
+ } else if (strncmp(cmd, "g", 1) == 0) {
+ _webrtc_get_state();
+
+ } else {
+ g_print("unknown menu \n");
+ }
+
+ } else if (len == 2) {
+ if (strncmp(cmd, "st", 2) == 0) {
+ g_menu_state = CURRENT_STATUS_SET_STUN_SERVER;
+
+ } else if (strncmp(cmd, "ss", 2) == 0) {
+ g_menu_state = CURRENT_STATUS_SETTING_SIGNALLING_SERVER;
+
+ } else if (strncmp(cmd, "px", 2) == 0) {
+ g_menu_state = CURRENT_STATUS_SETTING_PROXY;
+
+ } else {
+ g_print("unknown menu \n");
+ }
+
+ } else {
+ g_print("unknown menu \n");
+ }
+}
+
+void display_sub_basic()
+{
+ g_print("\n");
+ g_print("=========================================================================================\n");
+ g_print(" Native WebRTC Test (press q to quit) \n");
+ g_print("-----------------------------------------------------------------------------------------\n");
+ g_print("c. Create\t");
+ g_print("d. Destroy\n");
+ g_print("s. Start\t");
+ g_print("t. Stop\n");
+ g_print("a. Add media source\t");
+ g_print("r. Remove media source\n");
+ g_print("st. Set STUN server\n");
+ g_print("g. Get state\n");
+ g_print("----------------------------------- App. Setting ----------------------------------------\n");
+ g_print("ss. Signalling server\n");
+ g_print("px. Proxy\n");
+ g_print("-----------------------------------------------------------------------------------------\n");
+ g_print("=========================================================================================\n");
+}
+
+static void displaymenu()
+{
+ if (g_menu_state == CURRENT_STATUS_MAINMENU) {
+ display_sub_basic();
+ } else if (g_menu_state == CURRENT_STATUS_ADD_MEDIA_SOURCE) {
+ g_print("*** input media source type.(1:camera, 2:mic, 3:audiotest, 4:videotest)\n");
+ } else if (g_menu_state == CURRENT_STATUS_REMOVE_MEDIA_SOURCE) {
+ g_print("*** input media source id to remove.\n");
+ } else if (g_menu_state == CURRENT_STATUS_SET_STUN_SERVER) {
+ g_print("*** input STUN server address.\n");
+ } else if (g_menu_state == CURRENT_STATUS_SETTING_SIGNALLING_SERVER) {
+ g_print("*** input signalling server url.\n");
+ } else if (g_menu_state == CURRENT_STATUS_SETTING_PROXY) {
+ g_print("*** input proxy address.\n");
+ } else {
+ g_print("*** unknown status.\n");
+ quit_program();
+ }
+ g_print(" >>> ");
+}
+
+gboolean timeout_menu_display(void *data)
+{
+ displaymenu();
+ return FALSE;
+}
+
+gboolean timeout_quit_program(void *data)
+{
+ quit_program();
+ return FALSE;
+}
+
+void reset_menu_state(void)
+{
+ g_menu_state = CURRENT_STATUS_MAINMENU;
+}
+
+static void interpret(char *cmd)
+{
+ int value;
+
+ switch (g_menu_state) {
+ case CURRENT_STATUS_MAINMENU:
+ _interpret_main_menu(cmd);
+ break;
+ case CURRENT_STATUS_ADD_MEDIA_SOURCE: {
+ value = atoi(cmd);
+ _webrtc_add_media_source(value);
+ reset_menu_state();
+ break;
+ }
+ case CURRENT_STATUS_REMOVE_MEDIA_SOURCE: {
+ value = atoi(cmd);
+ _webrtc_remove_media_source(value);
+ reset_menu_state();
+ break;
+ }
+ case CURRENT_STATUS_SET_STUN_SERVER: {
+ _webrtc_set_stun_server(cmd);
+ reset_menu_state();
+ break;
+ }
+ case CURRENT_STATUS_SETTING_SIGNALLING_SERVER: {
+ _setting_signalling_server(cmd);
+ reset_menu_state();
+ break;
+ }
+ case CURRENT_STATUS_SETTING_PROXY: {
+ _setting_proxy(cmd);
+ reset_menu_state();
+ break;
+ }
+ }
+
+ g_timeout_add(100, timeout_menu_display, 0);
+}
+
+gboolean input(GIOChannel *channel)
+{
+ gchar buf[MAX_STRING_LEN];
+ gsize read;
+ GError *error = NULL;
+
+ g_io_channel_read_chars(channel, buf, MAX_STRING_LEN, &read, &error);
+ buf[read] = '\0';
+ g_strstrip(buf);
+ interpret(buf);
+
+ return TRUE;
+}
+
+int main(int argc, char *argv[])
+{
+ GIOChannel *stdin_channel;
+ stdin_channel = g_io_channel_unix_new(0);
+ g_io_channel_set_flags(stdin_channel, G_IO_FLAG_NONBLOCK, NULL);
+ g_io_add_watch(stdin_channel, G_IO_IN, (GIOFunc)input, NULL);
+
+ displaymenu();
+ memset(&ad, 0x0, sizeof(appdata));
+ ops.data = &ad;
+
+ return appcore_efl_main(PACKAGE, &argc, &argv, &ops);
+}
projects / platform / core / api / webrtc.git / commitdiff