I`m a beginner at using Gstreamer to handle some input videos. I have already built the pipeline using GStreamer to transcode the videos but the last part I cannot do is How I can get those batches of frames and do some custom image processing techniques to handle the purpose of my task.
Input Videos -----> Gstreamer Pipeline -----> Task: Apply some Image Processing Techniques
I`ve been searching about this problem on the Internet but cannot find any solution and the more I search, the more I am confused.
AppSink is the good element for you. You can enable "emit-signal" property and listen the event "new-sample". Then you can get an access to the buffer.
Here the entire documentation :
https://gstreamer.freedesktop.org/documentation/tutorials/basic/short-cutting-the-pipeline.html?gi-language=c
You have to create appsink element, enable "emit-signals" then register "new-sample" callback like this :
g_signal_connect (data.app_sink, "new-sample", G_CALLBACK (new_sample), &data)
static GstFlowReturn new_sample (GstElement *sink, CustomData *data) {
GstSample *sample;
/* Retrieve the buffer */
g_signal_emit_by_name (sink, "pull-sample", &sample);
if (sample) {
/* The only thing we do in this example is print a * to indicate a received buffer */
g_print ("*");
gst_sample_unref (sample);
return GST_FLOW_OK;
}
return GST_FLOW_ERROR;
}
Now you can retrieve buffer from sample instead of g_print ... (gst_sample_get_buffer)
https://gstreamer.freedesktop.org/documentation/gstreamer/gstsample.html?gi-language=c
Then read data inside the buffer :
GstMapInfo info;
gst_buffer_map (buf, &info, GST_MAP_READ);
gst_buffer_unmap (buf, &info);
gst_buffer_unref (buf);
info.data ==> buffer content.
Best regards.
Related
Is it possible to pass two streams through a single element? I have two streams
Need to extract data from, can be destroyed in element or passed
through to a sink.
Video stream, will be edited based on the
data extracted from stream 1, passed through to autovideosink
GStreamer Core Library version 1.16.2
Written in c
chain functions:
static GstFlowReturn
gst_test2_chain (GstPad * pad, GstObject * parent, GstBuffer * buf)
{
Gsttest2 *filter;
filter = GST_TEST2 (parent);
/* just push out the incoming buffer without touching it */
return gst_pad_push (filter->srcpad, buf);
}
//second pads chain function
static GstFlowReturn
gst_test2_chain2 (GstPad * pad, GstObject * parent, GstBuffer * buf)
{
g_print("\ninside chain2\n");
Gsttest2 *filter;
filter = GST_TEST2 (parent);
return gst_pad_push (filter->srcpad2, buf);
}
//Pad templates:
static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("video/x-raw")
);
Extracting the data from one stream and editing the other works fine. Currently using two video/x-raw src and sink pads for testing, but the stream for extracting the data would eventually be meta/x-klv. Using a single pad and source works fine with videotestsrc, but trying to use both sources and sinks result in pipeline errors unable to link or syntax. Does gstreamer support sending two streams through a single element? Would it be simpler to destroy the buffer of the no longer needed stream in element?
I'm working on a project in openGL and it needs to be able to play simple sounds (mp3) from file while not interrupting the draw loop.
I've been playing around with a few different libraries (openAL, portaudio) and eventually settled on mpg123 (to load the mp3) and libao to play the mp3 back.
The current playsound function works but it blocks the openGL draw loop (ie. freezes the game) until the audio has completed playing. I have tried messing around with std::thread but it still blocked the draw loop.
Here is the audio playback function I've been testing with:
void playSound() {
mpg123_handle *mh;
unsigned char *buffer;
size_t buffer_size;
size_t done;
int err;
int driver;
ao_device *dev;
ao_sample_format format;
int channels, encoding;
long rate;
/* initializations */
ao_initialize();
driver = ao_default_driver_id();
mpg123_init();
mh = mpg123_new(NULL, &err);
buffer_size = mpg123_outblock(mh);
buffer = (unsigned char*) malloc(buffer_size * sizeof(unsigned char));
/* open the file and get the decoding format */
mpg123_open(mh, "sounds/door.mp3");
mpg123_getformat(mh, &rate, &channels, &encoding);
/* set the output format and open the output device */
format.bits = mpg123_encsize(encoding) * 8;
format.rate = rate;
format.channels = channels;
format.byte_format = AO_FMT_NATIVE;
format.matrix = 0;
dev = ao_open_live(driver, &format, NULL);
/* decode and play */
while (mpg123_read(mh, buffer, buffer_size, &done) == MPG123_OK)
ao_play(dev, (char*)buffer, done);
/* clean up */
free(buffer);
ao_close(dev);
mpg123_close(mh);
mpg123_delete(mh);
mpg123_exit();
ao_shutdown();
}
How would I go about fixing this so that my game continues to run smoothly and the audio plays in the background?
You should unpack small amount of audio data and feed it to an audio device every frame.
The main trick is to find out how many samples was played by device already. I'm not sure how you can do this with libao, but it pretty simple with OpenAL.
You can check details here Play stream in OpenAL library
Also, you always can use additional thread. It'll be overkill, but very simple to do and can work fine for a small/demo project.
I have char* buffer that I read from video.mp4 file. This buffer has size 4096.
I tried to create GstBuffer from char* buffer
GstBuffer* Buffer = gst_buffer_new_wrapped(data, size);
dataBuffer = gst_buffer_copy(tmpBuf);
Then I push this buffer to the appsrc
GstElement* source = gst_bin_get_by_name (GST_BIN (consumer), "source");
gst_app_src_push_buffer (GST_APP_SRC (source), dataBuffer);
gst_object_unref (source);
Pipeline consumer was created in the next way:
gchar* videoConsumerString = g_strdup_printf ("appsrc max-buffers=5 drop=false name=source ! decodebin ! xvimagesink");
consumer = gst_parse_launch (videoConsumerString, NULL);
gst_element_set_state (consumer, GST_STATE_NULL);
g_free (videoConsumerString);
After the create of pipeline I set its state to the GST_STATE_NULL.
When I starts playing I set its state to GST_STATE_PLAYING.
But in the out I got error:
ERROR from element mpegvparse0: No valid frames found before end of stream
I tried to change size of char* buffer, use different elements in the pipeline (e.g. ffmpegcolorspace, videconvert, some other) but did not resolve this issue.
If run with GST_DEBUG=3, i have a lot of warnings
0:00:00.064480642 4059 0x12c66d0 WARN codecparsers_mpegvideo gstmpegvideoparser.c:887:gst_mpeg_video_packet_parse_picture_header: Unsupported picture type : 0
I use gstreamer 1.0.
Does anybody faced with such problem?
P.S. I don't have possibility to read data from file with Gstreamer. I only can read buffers from file with fread and then try to play them.
Maybe I have to set some specific fixed size of readed buffer?
I solved this problem.
Unexpectedly for me it was in the creating of the GstBuffer.
Correct way to create such buffer from data(char*) with known size is
GstBuffer * buffer = gst_buffer_new_allocate(NULL, size, NULL);
gst_buffer_fill(m_dataBufferProducer, 0, data, size);
Thank you for your help!
I need to mux klv metadata into the h264 stream. I have created application. But the stream is playing only as long as klv-data is being inserted. When i stop pushing klv-data the whole stream stops. What is the right method to mux asynchronous klv data by mpegtsmux?
Klv-data need to be inserted into the following working pipeline:
v4l2src input-src=Camera ! videorate drop-only=true ! 'video/x-raw, format=(string)NV12, width=1920, height=1088, framerate=25/1' ! ce_h264enc target-bitrate=6000000 idrinterval=25 intraframe-interval=60 ! queue ! mpegtsmux alignment=7 ! udpsink host=192.168.0.1 port=3000 -v
This pipeline is collected in the application. To insert klv-metedata appsrc is created:
appSrc = gst_element_factory_make("appsrc", nullptr);
gst_app_src_set_caps (GST_APP_SRC (appSrc), gst_caps_new_simple("meta/x-klv", "parsed", G_TYPE_BOOLEAN, TRUE, "sparse", G_TYPE_BOOLEAN, TRUE, nullptr));
g_object_set(appSrc, "format", GST_FORMAT_TIME, nullptr);
Then appsrc is linked to the pipeline:
gst_bin_add(GST_BIN(pipeline), appSrc);
gst_element_link(appSrc, mpegtsmux);
Here is push function:
void AppSrc::pushData(const std::string &data)
{
GstBuffer *buffer = gst_buffer_new_allocate(nullptr, data.size(), nullptr);
GstMapInfo map;
GstClock *clock;
GstClockTime abs_time, base_time;
gst_buffer_map (buffer, &map, GST_MAP_WRITE);
memcpy(map.data, data.data(), data.size());
gst_buffer_unmap (buffer, &map);
GST_OBJECT_LOCK (element);
clock = GST_ELEMENT_CLOCK (element);
base_time = GST_ELEMENT (element)->base_time;
gst_object_ref (clock);
GST_OBJECT_UNLOCK (element);
abs_time = gst_clock_get_time (clock);
gst_object_unref (clock);
GST_BUFFER_PTS (buffer) = abs_time - base_time;
GST_BUFFER_DURATION (buffer) = gst_util_uint64_scale_int (1, GST_SECOND, 1);
gst_app_src_push_buffer(GST_APP_SRC(element), buffer);
}
Gstreamer version is 1.6.1.
What can be wrong with my code? I'd appreciate your help.
I can push dummy klv-packets to maintain video stream. But i don't want to pollute upcomming stream and i am sure there should be more delicate solution.
I have found that i can send event with GST_STREAM_FLAG_SPARSE, which should be appropriate for subtitles. But as a result i have no output at all.
GstEvent* stream_start = gst_event_new_stream_start("klv-04");
gst_event_set_stream_flags(stream_start, GST_STREAM_FLAG_SPARSE);
GstPad* pad = gst_element_get_static_pad(GST_ELEMENT(element), "src");
gst_pad_push_event (pad, stream_start);
While debugging i have found that after applying the following patch to the gstreamer and using GST_STREAM_FLAG_SPARSE, the stream doesn't stop when the appsrc stops pushing packets.
diff --git a/libs/gst/base/gstcollectpads.c b/libs/gst/base/gstcollectpads.c
index 8edfe41..14f9926 100644
--- a/libs/gst/base/gstcollectpads.c
+++ b/libs/gst/base/gstcollectpads.c
## -1440,7 +1440,8 ## gst_collect_pads_recalculate_waiting (GstCollectPads * pads)
if (!GST_COLLECT_PADS_STATE_IS_SET (data, GST_COLLECT_PADS_STATE_WAITING)) {
/* start waiting */
gst_collect_pads_set_waiting (pads, data, TRUE);
- result = TRUE;
+ if (!GST_COLLECT_PADS_STATE_IS_SET (data, GST_COLLECT_PADS_STATE_LOCKED))
+ result = TRUE;
}
}
}
Anyway, the receiver stops updating screen 10 seconds after the last klv packet.
This is a bit of an old thread but,
In my experience though, if there is no queue between the appsrc and the muxer, you will get this behavior. I would change your:
gst_element_link(appSrc, mpegtsmux);
To this:
gst_element_link(appSrc, appSrcQueue);
gst_element_link(appSrcQueue, mpegtsmux);
And I'm not sure if the mpegtsmux has the capability for it or not but the muxer that we have used has a property called do-timestamping and when that was set to TRUE we had a better experience.
Another tip I would give is to use the gst-inspect tool to see what options each elements have.
I'm implementing gstreamer media player with my own source of data using appsrc. Everything works fine except one thing:
When stream reaches it's end, callback emits "end-of-stream" signal. Signals sending fucntion g_signal_emit_by_name(appsrc, "end-of-stream", &ret) returns GstFlowReturn value GST_FLOW_OK. But then it calls need-data my callback again, so it returns "end-of-stream" signal again. And this time GstFlowReturn value is (-3) which is GST_FLOW UNEXPECTED. I assume that it does not expect "end-of-stream" signal when it already recieved one, but why it requests more data than? Maybe it is because I didn't set size value iof the steam?
Gstreamer version is 0.10.
Callback function code (appsrc type is seekable btw):
static void cb_need_data (GstElement *appsrc, guint size, gpointer user_data)
{
GstBuffer *buffer;
GstFlowReturn ret;
AppsrcData* data = static_cast<AppsrcData*>(user_data);
buffer = gst_buffer_new_and_alloc(size);
int read = fread(GST_BUFFER_DATA(buffer), 1, size, data->file);
GST_BUFFER_SIZE(buffer) = read;
g_signal_emit_by_name (appsrc, "push-buffer", buffer, &ret);
if (ret != GST_FLOW_OK) {
/* something wrong, stop pushing */
g_printerr("GST_FLOW != OK, return value is %d\n", ret);
g_main_loop_quit (data->loop);
}
if(feof(data->file) || read == 0)
{
g_signal_emit_by_name(appsrc, "end-of-stream", &ret);
if (ret != GST_FLOW_OK) {
g_printerr("EOF reached, GST_FLOW != OK, return value is %d\nAborting...", ret);
g_main_loop_quit (data->loop);
}
}
}
You should provide some corrections to your code(if they are not there already) that should alleviate your issue and help the overall application:
Never try and send a buffer without first checking if it actually has data. So, simply check the buffer data and length to make sure that the data is not NULL and that the length is >0
You can flag that a stream is ended in your user_data. When you send your EOS, set an item in your userdata to indicate that it has been sent and if the appsrc requests more data, simply check if it has been sent and then do not send anything else to the buffer.
Listen for the EOS on your pipeline bus so that it can destroy the stream and close the loop when the EOS message is handled so that you can be sure that your mediasink has received the EOS and you can safely dispose of the pipeline and loop without losing any data.
Have you tried the method gst_app_src_end_of_stream()? I'm not sure what return code you should use after invoking it, but it should be either GST_FLOW_OK or GST_FLOW_UNEXPECTED.
In GStreamer 1.x you return GST_FLOW_EOS.