Name: capi-media-webrtc
Summary: A WebRTC library in Tizen Native API
-Version: 0.3.266
+Version: 0.3.267
Release: 0
Group: Multimedia/API
License: Apache-2.0
g_print("webrtc_get_display_visible() success, track_id[%u], visible[%u]\n", track_id, visible);
}
+static void _webrtc_set_audio_mute(int index, unsigned int track_id, int mute)
+{
+ int ret = webrtc_set_audio_mute(g_ad.conns[index].webrtc, track_id, (bool)mute);
+ RET_IF(ret != WEBRTC_ERROR_NONE, "ret[0x%x]", ret);
+
+ g_print("webrtc_set_audio_mute() success, track_id[%u], mute[%u]\n", track_id, mute);
+}
+
+static void _webrtc_get_audio_mute(int index, unsigned int track_id)
+{
+ int ret = WEBRTC_ERROR_NONE;
+ bool muted;
+
+ ret = webrtc_get_audio_mute(g_ad.conns[index].webrtc, track_id, &muted);
+ RET_IF(ret != WEBRTC_ERROR_NONE, "ret[0x%x]", ret);
+
+ g_print("webrtc_get_audio_mute() success, track_id[%u], muted[%u]\n", track_id, muted);
+}
+
+static void _webrtc_get_video_resolution(int index, unsigned int track_id)
+{
+ int ret = WEBRTC_ERROR_NONE;
+ int width;
+ int height;
+
+ ret = webrtc_get_video_resolution(g_ad.conns[index].webrtc, track_id, &width, &height);
+ RET_IF(ret != WEBRTC_ERROR_NONE, "ret[0x%x]", ret);
+
+ g_print("webrtc_get_video_resolution() success, track_id[%u], width[%d], height[%d]\n", track_id, width, height);
+}
+
static void _webrtc_media_source_set_audio_loopback(int index, unsigned int source_id)
{
int ret = WEBRTC_ERROR_NONE;
value = atoi(cmd);
_webrtc_get_display_visible(0, value);
break;
+ case CURRENT_STATUS_SET_AUDIO_MUTE: {
+ static unsigned int id;
+ value = atoi(cmd);
+
+ switch (g_ad.input_count) {
+ case 0:
+ id = value;
+ g_ad.input_count++;
+ return;
+ case 1:
+ _webrtc_set_audio_mute(0, id, value);
+ id = 0;
+ g_ad.input_count = 0;
+ break;
+ }
+ break;
+ }
+ case CURRENT_STATUS_GET_AUDIO_MUTE:
+ value = atoi(cmd);
+ _webrtc_get_audio_mute(0, value);
+ break;
+ case CURRENT_STATUS_GET_VIDEO_RESOLUTION:
+ value = atoi(cmd);
+ _webrtc_get_video_resolution(0, value);
+ break;
case CURRENT_STATUS_SET_ENCODED_AUDIO_FRAME_CB:
_webrtc_set_encoded_audio_frame_cb(0);
break;
{ "gm", CURRENT_STATUS_GET_DISPLAY_MODE, true },
{ "dv", CURRENT_STATUS_SET_DISPLAY_VISIBLE, true },
{ "gv", CURRENT_STATUS_GET_DISPLAY_VISIBLE, true },
+ { "sam", CURRENT_STATUS_SET_AUDIO_MUTE, true },
+ { "gam", CURRENT_STATUS_GET_AUDIO_MUTE, true },
+ { "gvr", CURRENT_STATUS_GET_VIDEO_RESOLUTION, true },
{ "sa", CURRENT_STATUS_SET_ENCODED_AUDIO_FRAME_CB, false },
{ "ua", CURRENT_STATUS_UNSET_ENCODED_AUDIO_FRAME_CB, false },
{ "sv", CURRENT_STATUS_SET_ENCODED_VIDEO_FRAME_CB, false },
g_print("gm. Get display mode\n");
g_print("dv. Set display visible\t");
g_print("gv. Get display visible\n");
+ g_print("*sam. Set audio mute\t");
+ g_print("*gam. Get audio mute\n");
+ g_print("*gvr. Get video resolution\n");
g_print("al. Set audio loopback\t");
g_print("ual. Unset audio loopback\n");
g_print("vl. Set video loopback\t");
case CURRENT_STATUS_GET_DISPLAY_VISIBLE:
g_print("*** input track id.\n");
break;
+ case CURRENT_STATUS_SET_AUDIO_MUTE:
+ if (get_appdata()->input_count == 0)
+ g_print("*** input track id.\n");
+ else if (get_appdata()->input_count == 1)
+ g_print("*** input audio mute.(1:true 0:false)\n");
+ break;
+ case CURRENT_STATUS_GET_AUDIO_MUTE:
+ g_print("*** input track id.\n");
+ break;
+ case CURRENT_STATUS_GET_VIDEO_RESOLUTION:
+ g_print("*** input track id.\n");
+ break;
case CURRENT_STATUS_MEDIA_SOURCE_SET_AUDIO_LOOPBACK:
g_print("*** input source id.\n");
break;
CURRENT_STATUS_GET_DISPLAY_MODE = TEST_MENU_WEBRTC_MEDIA_RENDER | 0x03,
CURRENT_STATUS_SET_DISPLAY_VISIBLE = TEST_MENU_WEBRTC_MEDIA_RENDER | 0x04,
CURRENT_STATUS_GET_DISPLAY_VISIBLE = TEST_MENU_WEBRTC_MEDIA_RENDER | 0x05,
- CURRENT_STATUS_SET_ENCODED_AUDIO_FRAME_CB = TEST_MENU_WEBRTC_MEDIA_RENDER | 0x06,
- CURRENT_STATUS_UNSET_ENCODED_AUDIO_FRAME_CB = TEST_MENU_WEBRTC_MEDIA_RENDER | 0x07,
- CURRENT_STATUS_SET_ENCODED_VIDEO_FRAME_CB = TEST_MENU_WEBRTC_MEDIA_RENDER | 0x08,
- CURRENT_STATUS_UNSET_ENCODED_VIDEO_FRAME_CB = TEST_MENU_WEBRTC_MEDIA_RENDER | 0x09,
- CURRENT_STATUS_MEDIA_SOURCE_SET_AUDIO_LOOPBACK = TEST_MENU_WEBRTC_MEDIA_RENDER | 0x0A,
- CURRENT_STATUS_MEDIA_SOURCE_UNSET_AUDIO_LOOPBACK = TEST_MENU_WEBRTC_MEDIA_RENDER | 0x0B,
- CURRENT_STATUS_MEDIA_SOURCE_SET_VIDEO_LOOPBACK = TEST_MENU_WEBRTC_MEDIA_RENDER | 0x0C,
- CURRENT_STATUS_MEDIA_SOURCE_UNSET_VIDEO_LOOPBACK = TEST_MENU_WEBRTC_MEDIA_RENDER | 0x0D,
+ CURRENT_STATUS_SET_AUDIO_MUTE = TEST_MENU_WEBRTC_MEDIA_RENDER | 0x06,
+ CURRENT_STATUS_GET_AUDIO_MUTE = TEST_MENU_WEBRTC_MEDIA_RENDER | 0x07,
+ CURRENT_STATUS_GET_VIDEO_RESOLUTION = TEST_MENU_WEBRTC_MEDIA_RENDER | 0x08,
+ CURRENT_STATUS_SET_ENCODED_AUDIO_FRAME_CB = TEST_MENU_WEBRTC_MEDIA_RENDER | 0x09,
+ CURRENT_STATUS_UNSET_ENCODED_AUDIO_FRAME_CB = TEST_MENU_WEBRTC_MEDIA_RENDER | 0x0A,
+ CURRENT_STATUS_SET_ENCODED_VIDEO_FRAME_CB = TEST_MENU_WEBRTC_MEDIA_RENDER | 0x0B,
+ CURRENT_STATUS_UNSET_ENCODED_VIDEO_FRAME_CB = TEST_MENU_WEBRTC_MEDIA_RENDER | 0x0C,
+ CURRENT_STATUS_MEDIA_SOURCE_SET_AUDIO_LOOPBACK = TEST_MENU_WEBRTC_MEDIA_RENDER | 0x0D,
+ CURRENT_STATUS_MEDIA_SOURCE_UNSET_AUDIO_LOOPBACK = TEST_MENU_WEBRTC_MEDIA_RENDER | 0x0E,
+ CURRENT_STATUS_MEDIA_SOURCE_SET_VIDEO_LOOPBACK = TEST_MENU_WEBRTC_MEDIA_RENDER | 0x0F,
+ CURRENT_STATUS_MEDIA_SOURCE_UNSET_VIDEO_LOOPBACK = TEST_MENU_WEBRTC_MEDIA_RENDER | 0x10,
/* webrtc data channel */
CURRENT_STATUS_DATA_CHANNEL_CREATE = TEST_MENU_WEBRTC_DATA_CHANNEL | 0x01,
CURRENT_STATUS_DATA_CHANNEL_DESTROY = TEST_MENU_WEBRTC_DATA_CHANNEL | 0x02,