Gstreamer RTSP src element name
rajvik
kamdar.rajvi at gmail.com
Tue Jan 24 14:45:43 UTC 2017
It gets into the ifelse condition. Still facing error:
name of caps string: application/x-rtp, media=(string)video,
payload=(int)96, clock-rate=(int)90000, encoding-name=(string)H264,
packetization-mode=(string)1, profile-level-id=(string)42801e,
sprop-parameter-sets=(str
ing)"Z0KAHpZWDY/yf/gACAAKhAAAD6QAA6mDgAAC3GAAFuNvxjg7QsXc\,aMqNSA\=\=",
a-tool=(string)"vlc\ 2.1.6", a-recvonly=(string)"",
a-type=(string)broadcast, a-charset=(string)UTF-8, ssrc=(uint)3197239660,
clock-base=(uint)22
81054233, seqnum-base=(uint)15975, npt-start=(guint64)7813930284000,
play-speed=(double)1, play-scale=(double)1
name of caps struct string: application/x-rtp
name of caps string: application/x-rtp, media=(string)audio,
payload=(int)96, clock-rate=(int)44100,
encoding-name=(string)MPEG4-GENERIC, encoding-params=(string)2,
streamtype=(string)5, profile-level-id=(string)15, m
ode=(string)AAC-hbr, config=(string)1210, sizelength=(string)13,
indexlength=(string)3, indexdeltalength=(string)3, profile=(string)1,
a-tool=(string)"vlc\ 2.1.6", a-recvonly=(string)"",
a-type=(string)broadcast, a-ch
arset=(string)UTF-8, ssrc=(uint)358833443, clock-base=(uint)1031817228,
seqnum-base=(uint)16576, npt-start=(guint64)7813930284000,
play-speed=(double)1, play-scale=(double)1
0:00:00.795704054 2728 0x188430 WARN basesrc
gstbasesrc.c:2933:gst_base_src_loop:<udpsrc1> error: Internal data flow
error.
0:00:00.812515845 2728 0x188430 WARN basesrc
gstbasesrc.c:2933:gst_base_src_loop:<udpsrc1> error: streaming task paused,
reason not-linked (-1)
Any idea as to how to move forward with this?
Code snippet:
GstElement *audioQueue, *videoQueue;
GstElement *source, *audio, *video, *convert, *pipeline, *audioDepay,
*audioParse, *audioDecode, *audioConvert, *audioResample,
*audioSink, *videoDepay, *videoParser, *videoDecode, *videoConvert,
*videoScale, *videoSink;
static void onPadAdded(GstElement *element, GstPad *pad, gpointer data)
{
GstCaps *caps;
const char *name;
char *capsName;
caps = gst_pad_get_current_caps(pad);
GstStructure *str = gst_caps_get_structure(caps, 0);
name = gst_structure_get_name(str);
g_print("name of caps struct string: %s \n", name);
capsName = gst_caps_to_string(caps);
g_print("name of caps string: %s \n", capsName);
if (g_strrstr(capsName,"audio"))
{
GstPad *dynamic_pad =
gst_element_get_static_pad(audioQueue, "sink");
gst_pad_link(pad, dynamic_pad);
}
else if (g_strrstr(capsName, "video"))
{
GstPad *video_dynamic_pad =
gst_element_get_static_pad(videoQueue, "sink");
gst_pad_link(pad, video_dynamic_pad);
}
g_free(capsName);
}
int main(int argc, char *argv[]) {
GstCaps *capsFilter;
GstBus *bus;
GstMessage *msg;
GstPad *pad;
gboolean link_ok;
GstStateChangeReturn ret;
/* Initialize GStreamer */
gst_init (&argc, &argv);
/* Create Elements */
pipeline = gst_pipeline_new("rtsp-pipeline");
source = gst_element_factory_make ("rtspsrc", "rtsp-source");
g_object_set(source, "location", "rtsp://192.168.3.30:8554/rajvi",
NULL);
g_object_set(source, "latency", 0, NULL);
audioQueue = gst_element_factory_make ("queue", "audio-queue");
audioDepay = gst_element_factory_make ("rtpmp4gdepay",
"audio-depayer");
audioParse = gst_element_factory_make ("aacparse", "audio-parser");
audioDecode = gst_element_factory_make ("avdec_aac",
"audio-decoder");
audioConvert = gst_element_factory_make ("audioconvert", "aconv");
audioResample = gst_element_factory_make ("audioresample",
"audio-resample");
audioSink = gst_element_factory_make ("autoaudiosink", "audiosink");
if (!audioQueue || !audioDepay || !audioParse || !audioConvert ||
!audioResample || !audioSink)
{
g_printerr("Cannot create audio elements \n");
return 0;
}
videoQueue = gst_element_factory_make ("queue", "video-queue");
videoDepay= gst_element_factory_make ("rtph264depay",
"video-depay");
videoParser = gst_element_factory_make ("h264parse",
"video-parser");
videoDecode = gst_element_factory_make ("omxh264dec",
"video-decoder");
videoConvert = gst_element_factory_make("videoconvert", "convert");
videoScale = gst_element_factory_make("videoscale", "video-scale");
videoSink = gst_element_factory_make("ximagesink", "video-sink");
capsFilter = gst_caps_new_simple("video/x-raw",
"width", G_TYPE_INT, 176,
"height", G_TYPE_INT, 144,
NULL);
if (!videoQueue || !videoDepay || !videoParser || !videoDecode ||
!videoConvert || !videoScale || !videoSink || !capsFilter)
{
g_printerr("Cannot create video elements \n");
return 0;
}
gst_bin_add(GST_BIN(pipeline), source);
gst_bin_add_many(GST_BIN(pipeline),
audioQueue, audioDepay, audioParse, audioDecode,
audioConvert, audioResample, audioSink,
videoQueue, videoDepay, videoParser, videoDecode,
videoConvert, videoScale, videoSink, NULL);
if (!gst_element_link_many(audioQueue, audioDepay, NULL ))
{
g_printerr("Cannot link audioqueue and audioDepay \n");
return 0;
}
if (!gst_element_link_many(audioParse, audioDecode, NULL ))
{
g_printerr("Cannot link audioParse and audiodecode \n");
return 0;
}
if (!gst_element_link_many( audioConvert, audioResample, audioSink,
NULL ))
{
g_printerr("Cannot link audioConvert, audioParse, audioSink
\n");
return 0;
}
/*Linking filter element to videoScale and videoSink */
link_ok = gst_element_link_filtered(videoScale,videoSink,
capsFilter);
gst_caps_unref (capsFilter);
if (!link_ok) {
g_warning ("Failed to link element1 and element2!");
return 0;
}
/* Linking video elements internally */
if (!gst_element_link_many(videoQueue, videoDepay, NULL))
{
g_printerr("Cannot link videoQueue and videoDepay \n");
return 0;
}
if (!gst_element_link_many(videoParser, videoDecode, videoConvert,
NULL))
{
g_printerr("Cannot link videoParser, videoDecode,
videoConvert \n");
return 0;
}
g_signal_connect(source, "pad-added", G_CALLBACK(onPadAdded), NULL);
/* Start playing */
gst_element_set_state ( pipeline, GST_STATE_PLAYING);
/* Wait until error or EOS */
bus = gst_element_get_bus (pipeline);
msg = gst_bus_timed_pop_filtered (bus, GST_CLOCK_TIME_NONE,
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);
return 0;
}
On Tue, Jan 24, 2017 at 6:29 PM, Tarun Tej K [via GStreamer-devel] <
ml-node+s966125n4681617h35 at n4.nabble.com> wrote:
> Try g_strstr instead of g_str_has_prefix
>
>
> On 24-Jan-2017 6:16 PM, "rajvik" <[hidden email]
> <http:///user/SendEmail.jtp?type=node&node=4681617&i=0>> wrote:
>
> The problem is it is not getting into the if else statement.
>
> if (g_str_has_prefix(capsName,"audio"))
> {
> g_print("Here 6th .....\n");
> GstPad *dynamic_pad = gst_element_get_static_pad(audioQueue,
> "sink");
> gst_pad_link(pad, dynamic_pad);
> }
>
> else if (g_str_has_prefix(capsName, "video"))
> {
> g_print("Here 7th .....\n");
> GstPad *video_dynamic_pad = gst_element_get_static_pad(videoQueue,
> "sink");
> gst_pad_link(pad, video_dynamic_pad);
> }
> g_free(capsName);
>
> Reason being the prefix of capsName is :
> name of caps string: application/x-rtp, media=(string)audio,
> payload=(int)96, clock-rate=(int)44100, encoding-name=(string)MPEG4-GENERIC,
> encoding-params=(string)2, streamtype=(string)5,
> profile-level-id=(string)15, m
> ode=(string)AAC-hbr, config=(string)1210, sizelength=(string)13,
> indexlength=(string)3, indexdeltalength=(string)3, profile=(string)1,
> a-tool=(string)"vlc\ 2.1.6", a-recvonly=(string)"",
> a-type=(string)broadcast, a-ch
> arset=(string)UTF-8, ssrc=(uint)398000458, clock-base=(uint)687408962,
> seqnum-base=(uint)15402, npt-start=(guint64)4219024000,
> play-speed=(double)1, play-scale=(double)1
>
> Any API which would do a string compare for 1 word from the whole string?
>
> On Tue, Jan 24, 2017 at 5:14 PM, Tarun Tej K [via GStreamer-devel] <[hidden
> email] <http:///user/SendEmail.jtp?type=node&node=4681616&i=0>> wrote:
>
>> I am not sure, but can you try without having the queues? Can you please
>> see how it works when link the pads with audioDepay and videoDepay directly?
>>
>> Tarun
>>
>> On 24-Jan-2017 4:31 PM, "rajvik" <[hidden email]
>> <http:///user/SendEmail.jtp?type=node&node=4681615&i=0>> wrote:
>>
>> Thank you so much for the insights. I did incorporate what you asked me
>> to,
>> but still I am getting error:
>> *0:00:00.602211533 2055 0x188380 WARN basesrc
>> gstbasesrc.c:2933:gst_base_src_loop:<udpsrc3> error: Internal data flow
>> error.
>> 0:00:00.618291500 2055 0x188380 WARN basesrc
>> gstbasesrc.c:2933:gst_base_src_loop:<udpsrc3> error: streaming task
>> paused,
>> reason not-linked (-1)
>> 0:00:00.638866649 2055 0x188430 WARN basesrc
>> gstbasesrc.c:2933:gst_base_src_loop:<udpsrc0> error: Internal data flow
>> error.
>> 0:00:00.655050073 2055 0x188430 WARN basesrc
>> gstbasesrc.c:2933:gst_base_src_loop:<udpsrc0> error: streaming task
>> paused,
>> reason not-linked (-1)*
>>
>> Please find the below code snippet:
>>
>> *Taken audioQueue and VideoQueue as global variables and find the below cb
>> function: *
>>
>> static void onPadAdded(GstElement *element, GstPad *pad, gpointer data)
>> {
>>
>> GstCaps *caps;
>> const char *name;
>> char *capsName;
>> caps = gst_pad_get_current_caps(pad);
>> GstStructure *str = gst_caps_get_structure(caps, 0);
>> name = gst_structure_get_name(str);
>> g_debug("name of caps struct string: %s", name);
>> capsName = gst_caps_to_string(caps);
>> g_debug("name of caps string: %s", capsName);
>>
>> if (g_str_has_prefix(capsName,"audio"))
>> {
>> GstPad *dynamic_pad = gst_element_get_static_pad(aud
>> ioQueue,
>> "sink");
>> gst_pad_link(pad, dynamic_pad);
>> }
>>
>> else if (g_str_has_prefix(capsName, "video"))
>> {
>> GstPad *video_dynamic_pad =
>> gst_element_get_static_pad(videoQueue, "sink");
>> gst_pad_link(pad, video_dynamic_pad);
>> }
>> g_free(capsName);
>>
>> }
>>
>> *Following is the g_signal_connect operation:*
>>
>>
>> gst_bin_add(GST_BIN(pipeline), source);
>> gst_bin_add_many(GST_BIN(pipeline),
>> audioQueue, audioDepay, audioParse, audioDecode,
>> audioConvert, audioResample, audioSink,
>> videoQueue, videoDepay, videoParser, videoDecode,
>> videoConvert, videoScale, videoSink, NULL);
>>
>> if (!gst_element_link_many(audioQueue, audioDepay, NULL ))
>> {
>> g_printerr("Cannot link audioqueue and audioDepay \n");
>> return 0;
>> }
>> if (!gst_element_link_many(audioParse, audioDecode,NULL ))
>> {
>> g_printerr("Cannot link audioParse and audiodecode \n");
>> return 0;
>> }
>> if (!gst_element_link_many(audioConvert, audioResample,
>> audioSink,
>> NULL ))
>> {
>> g_printerr("Cannot link audioConvert, audioParse,
>> audioSink
>> \n");
>> return 0;
>> }
>>
>>
>> /*Linking filter element to videoScale and videoSink */
>> link_ok = gst_element_link_filtered(videoScale,videoSink,
>> capsFilter);
>> gst_caps_unref (capsFilter);
>> if (!link_ok) {
>> g_warning ("Failed to link element1 and element2!");
>> }
>> /* Linking video elements internally */
>> if (!gst_element_link_many(videoQueue, videoDepay, NULL))
>> {
>> g_printerr("Cannot link videoQueue and videoDepay \n");
>> return 0;
>> }
>> if (!gst_element_link_many( videoParser, videoDecode,
>> videoConvert,
>> NULL))
>> {
>> g_printerr("Cannot link videoParser, videoDecode,
>> videoConvert \n");
>> return 0;
>> }
>> g_signal_connect(source, "pad-added", G_CALLBACK(onPadAdded),
>> NULL);
>>
>>
>>
>>
>> --
>> View this message in context: http://gstreamer-devel.966125.
>> n4.nabble.com/Gstreamer-RTSP-src-element-name-tp4681595p4681613.html
>>
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