problom with constructing GstBuffer

[Ugly Face]

I am developping a android program. I need to push NV21 raw video data into
pipeline which will transmit it with rtp. I use appsrc .

code is like this:

static void gst_native_push_data(JNIEnv * env, jobject thiz, jbyteArray
fdata)
{
	int data_len = 115200;
	int frame_rate = 15;
	GstBuffer *buffer;
	buffer = gst_buffer_new_and_alloc(data_len);
	CustomData *data = GET_CUSTOM_DATA (env, thiz, custom_data_field_id);

	GST_BUFFER_TIMESTAMP (buffer) = gst_util_uint64_scale (data->frame_num,
GST_SECOND, frame_rate);
	GST_BUFFER_DURATION (buffer) = gst_util_uint64_scale (1, GST_SECOND,
frame_rate);

	data->frame_num++;
	GST_BUFFER_DATA (buffer) = (*env)->GetByteArrayElements(env, fdata, 
JNI_FALSE);
	gst_app_src_push_buffer(data->appsrc, buffer);
	gst_buffer_unref(buffer);
}
with error like this :gstvideorate.c:830:gst_video_rate_event:<videorate>
Got segment but doesn't have GST_FORMAT_TIME value

I think I did not construct a Gstbuffer properly and I don't kwon what dose
this message mean.

code:
static void *app_function(CustomData *userdata)
{
	gboolean res;
  	GstPadLinkReturn linkres;
	GstCaps	   *caps;
	CustomData *data = (CustomData *)userdata;

	GstPad	   *srcpad;
	GstPad	   *sinkpad;

	GstElement *rtpbin;		// rtpbin to get rtp work.
	GstElement *queue;
	GstElement *videorate;	// to deal with timestamps.
	GstElement *videoconv;
	GstElement *x264encoder;// encode the raw data.
	GstElement *rtph264pay; // packetize the encoded data.
	GstElement *rtpsink;
	GstElement *rtcpsink;
	GstElement *rtcpsrc;
	gst_init(NULL, NULL);
	data->pipeline 	= gst_pipeline_new(NULL);
	g_assert(data->pipeline);

	data->appsrc= gst_element_factory_make("appsrc", "appsrc");
	g_assert(data->appsrc);
	g_object_set(data->appsrc, "is-live", TRUE, NULL);
	caps		= gst_caps_from_string(CAPS);
	g_object_set(data->appsrc, "caps", caps, NULL);
	
	queue		= gst_element_factory_make("queue", "queue");
	g_assert(queue);

	videorate	= gst_element_factory_make("videorate", "videorate");
	g_assert(videorate);

	videoconv	= gst_element_factory_make("ffmpegcolorspace", "videoconv");
	g_assert(videoconv);

	x264encoder = gst_element_factory_make("x264enc", "x264encoder");
	g_assert(x264encoder);
	g_object_set(x264encoder, "byte-stream", TRUE, "bitrate",
300,"speed-preset", 2, NULL);

	rtph264pay	= gst_element_factory_make("rtph264pay", "rtph264pay");
	g_assert(rtph264pay);
	g_object_set(rtph264pay, "config-interval", 2, NULL);

	rtpsink		= gst_element_factory_make("udpsink", "rtpsink");
	g_assert(rtpsink);
	g_object_set(rtpsink, "port", RTPPORT, "host", REMOTEIP,"ts-offset", 0,
NULL);

	rtcpsink	= gst_element_factory_make("udpsink", "rtcpsink");
	g_assert(rtcpsink);
	g_object_set(rtcpsink, "port", RTCPOUTPORT, "host", REMOTEIP, "sync",
FALSE, "async", FALSE, NULL);

	rtcpsrc		= gst_element_factory_make("udpsrc", "rtcpsrc");
	g_assert(rtcpsrc);
	g_object_set(rtcpsrc, "port", RTCPINPORT, NULL);

	rtpbin 		= gst_element_factory_make("gstrtpbin", "rtpbin");
	g_assert(rtpbin);

	__android_log_print (ANDROID_LOG_ERROR, "camera_gst_codec", "elements make
finished!");

	gst_bin_add_many(GST_BIN(data->pipeline), data->appsrc, queue, videorate,
videoconv, x264encoder,rtph264pay, 
		rtpsink, rtcpsink, rtcpsrc, rtpbin, NULL);

	res=gst_element_link(data->appsrc, queue);
	g_assert(res);
	
	res=gst_element_link(queue, videorate);
	g_assert(res);

	res=gst_element_link(videorate, videoconv);
	g_assert(res);

	res=gst_element_link(videoconv, x264encoder);
	g_assert(res);

	res=gst_element_link(x264encoder, rtph264pay);
	g_assert(res);

	srcpad 	= gst_element_get_static_pad(rtph264pay, "src");
	sinkpad	= gst_element_get_request_pad(rtpbin, "send_rtp_sink_0");
	linkres	= gst_pad_link(srcpad, sinkpad);
	g_assert(GST_PAD_LINK_OK == linkres);
	g_object_unref(srcpad);

	// lead rtp stream out of rtpbin via udpsink
	srcpad 	= gst_element_get_static_pad(rtpbin, "send_rtp_src_0");
	sinkpad	= gst_element_get_static_pad(rtpsink, "sink");
	linkres	= gst_pad_link(srcpad, sinkpad);
	g_assert(GST_PAD_LINK_OK == linkres);
	g_object_unref(srcpad);
	g_object_unref(sinkpad);

	// let rtcp stream into rtpbin via udpsrc.
	srcpad 	= gst_element_get_static_pad(rtcpsrc, "src");
	sinkpad	= gst_element_get_request_pad(rtpbin, "recv_rtcp_sink_0");
	linkres	= gst_pad_link(srcpad, sinkpad);
	g_assert(GST_PAD_LINK_OK == linkres);
	g_object_unref(srcpad);

	// lead rtcp stream out of rtpbin via udpsink.
	srcpad 	= gst_element_get_request_pad(rtpbin, "send_rtcp_src_0");
	sinkpad	= gst_element_get_static_pad(rtcpsink, "sink");
	linkres	= gst_pad_link(srcpad, sinkpad);
	g_assert(GST_PAD_LINK_OK == linkres);
	g_object_unref(sinkpad);

	g_print ("starting sender pipeline\n");
	gst_element_set_state (data->pipeline, GST_STATE_PLAYING);

	data->main_loop = g_main_loop_new (NULL, FALSE);
	g_main_loop_run (data->main_loop);

  	gst_element_set_state (data->pipeline, GST_STATE_NULL);
  	gst_object_unref(GST_OBJECT(data->pipeline));
}

with warning log like this:
01-03 21:51:03.750: W/GStreamer+bin(3310): 0:00:11.178009032 0x58faf9a0
gstbin.c:2399:gst_bin_do_latency_func:<pipeline0> failed to query latency
I do not know what is the meaning od this .

at last , the program fail with :
01-03 21:54:08.414: A/libc(3310): Fatal signal 11 (SIGSEGV) at 0x00000018
(code=1), thread 3332 (appsrc:src)


Thanks in advance.

Gorge




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