How can i write a pipeline that streams videotestsrc h265 encoded on RTSP and another that playback the former?
As far as i understand, this should be a valid server
gst-launch-1.0 -v videotestsrc ! video/x-raw,width=1280,height=720 ! x265enc ! rtph265pay ! udpsink host=127.0.0.1 port=5000
The output is
Setting pipeline to PAUSED ...
Pipeline is PREROLLING ...
/GstPipeline:pipeline0/GstVideoTestSrc:videotestsrc0.GstPad:src: caps = "video/x-raw\,\ format\=\(string\)I420\,\ width\=\(int\)1280\,\ height\=\(int\)720\,\ framerate\=\(fraction\)30/1\,\ pixel-aspect-ratio\=\(fraction\)1/1\,\ interlace-mode\=\(string\)progressive"
/GstPipeline:pipeline0/GstCapsFilter:capsfilter0.GstPad:src: caps = "video/x-raw\,\ format\=\(string\)I420\,\ width\=\(int\)1280\,\ height\=\(int\)720\,\ framerate\=\(fraction\)30/1\,\ pixel-aspect-ratio\=\(fraction\)1/1\,\ interlace-mode\=\(string\)progressive"
/GstPipeline:pipeline0/GstX265Enc:x265enc0.GstPad:sink: caps = "video/x-raw\,\ format\=\(string\)I420\,\ width\=\(int\)1280\,\ height\=\(int\)720\,\ framerate\=\(fraction\)30/1\,\ pixel-aspect-ratio\=\(fraction\)1/1\,\ interlace-mode\=\(string\)progressive"
Redistribute latency...
/GstPipeline:pipeline0/GstCapsFilter:capsfilter0.GstPad:sink: caps = "video/x-raw\,\ format\=\(string\)I420\,\ width\=\(int\)1280\,\ height\=\(int\)720\,\ framerate\=\(fraction\)30/1\,\ pixel-aspect-ratio\=\(fraction\)1/1\,\ interlace-mode\=\(string\)progressive"
/GstPipeline:pipeline0/GstX265Enc:x265enc0.GstPad:src: caps = "video/x-h265\,\ stream-format\=\(string\)byte-stream\,\ alignment\=\(string\)au\,\ level\=\(string\)3.1\,\ tier\=\(string\)main\,\ profile\=\(string\)main\,\ width\=\(int\)1280\,\ height\=\(int\)720\,\ pixel-aspect-ratio\=\(fraction\)1/1\,\ framerate\=\(fraction\)30/1"
/GstPipeline:pipeline0/GstRtpH265Pay:rtph265pay0.GstPad:sink: caps = "video/x-h265\,\ stream-format\=\(string\)byte-stream\,\ alignment\=\(string\)au\,\ level\=\(string\)3.1\,\ tier\=\(string\)main\,\ profile\=\(string\)main\,\ width\=\(int\)1280\,\ height\=\(int\)720\,\ pixel-aspect-ratio\=\(fraction\)1/1\,\ framerate\=\(fraction\)30/1"
/GstPipeline:pipeline0/GstRtpH265Pay:rtph265pay0.GstPad:src: caps = "application/x-rtp\,\ media\=\(string\)video\,\ payload\=\(int\)96\,\ clock-rate\=\(int\)90000\,\ encoding-name\=\(string\)H265\,\ ssrc\=\(uint\)2573237941\,\ timestamp-offset\=\(uint\)1713951204\,\ seqnum-offset\=\(uint\)27727\,\ a-framerate\=\(string\)30"
/GstPipeline:pipeline0/GstUDPSink:udpsink0.GstPad:sink: caps = "application/x-rtp\,\ media\=\(string\)video\,\ payload\=\(int\)96\,\ clock-rate\=\(int\)90000\,\ encoding-name\=\(string\)H265\,\ ssrc\=\(uint\)2573237941\,\ timestamp-offset\=\(uint\)1713951204\,\ seqnum-offset\=\(uint\)27727\,\ a-framerate\=\(string\)30"
/GstPipeline:pipeline0/GstRtpH265Pay:rtph265pay0.GstPad:src: caps = "application/x-rtp\,\ media\=\(string\)video\,\ clock-rate\=\(int\)90000\,\ encoding-name\=\(string\)H265\,\ sprop-parameter-sets\=\(string\)\"QAEMAf//AWAAAAMAkAAAAwAAAwBdlZgJAA\\\=\\\=\\\,QgEBAWAAAAMAkAAAAwAAAwBdoAKAgC0WWVmkkyuAQAAAAwBAAAAHggA\\\=\\\,RAHBcrRiQAA\\\=\"\,\ payload\=\(int\)96\,\ seqnum-offset\=\(uint\)27727\,\ timestamp-offset\=\(uint\)1713951204\,\ ssrc\=\(uint\)2573237941\,\ a-framerate\=\(string\)30"
/GstPipeline:pipeline0/GstUDPSink:udpsink0.GstPad:sink: caps = "application/x-rtp\,\ media\=\(string\)video\,\ clock-rate\=\(int\)90000\,\ encoding-name\=\(string\)H265\,\ sprop-parameter-sets\=\(string\)\"QAEMAf//AWAAAAMAkAAAAwAAAwBdlZgJAA\\\=\\\=\\\,QgEBAWAAAAMAkAAAAwAAAwBdoAKAgC0WWVmkkyuAQAAAAwBAAAAHggA\\\=\\\,RAHBcrRiQAA\\\=\"\,\ payload\=\(int\)96\,\ seqnum-offset\=\(uint\)27727\,\ timestamp-offset\=\(uint\)1713951204\,\ ssrc\=\(uint\)2573237941\,\ a-framerate\=\(string\)30"
/GstPipeline:pipeline0/GstRtpH265Pay:rtph265pay0: timestamp = 1713951204
/GstPipeline:pipeline0/GstRtpH265Pay:rtph265pay0: seqnum = 27727
Pipeline is PREROLLED ...
Setting pipeline to PLAYING ...
New clock: GstSystemClock
So i tried
gst-launch-1.0 udpsrc uri=udp://127.0.0.1:5000 caps="application/x-rtp, media=(string)video, clock-rate=(int)90000, encoding-name=(string)H265, payload=(int)96, seqnum-offset=(uint)27727, timestamp-offset=(uint)1713951204, ssrc=(uint)2573237941, a-framerate=(string)30" ! rtph265depay ! vaapidecode ! vaapipostproc ! vaapisink
but with no luck
libva info: VA-API version 0.39.3
libva info: va_getDriverName() returns 0
libva info: Trying to open /usr/lib/dri/i965_drv_video.so
libva info: Found init function __vaDriverInit_0_39
libva info: va_openDriver() returns 0
Setting pipeline to PAUSED ...
Pipeline is live and does not need PREROLL ...
Got context from element 'pipeline0': gst.vaapi.Display=context, gst.vaapi.Display=(GstVaapiDisplay)NULL;
Setting pipeline to PLAYING ...
New clock: GstSystemClock
ERROR: from element /GstPipeline:pipeline0/GstVaapiDecode:vaapidecode0: No valid frames decoded before end of stream
Additional debug info:
gstvideodecoder.c(1167): gst_video_decoder_sink_event_default (): /GstPipeline:pipeline0/GstVaapiDecode:vaapidecode0:
no valid frames found
Execution ended after 0:00:00.025823038
Setting pipeline to PAUSED ...
Setting pipeline to READY ...
Setting pipeline to NULL ...
Freeing pipeline ...
Try adding the parser before the decode like: gst-launch-1.0 udpsrc uri=udp://127.0.0.1:5000 caps="application/x-rtp, media=(string)video, clock-rate=(int)90000, encoding-name=(string)H265, payload=(int)96, seqnum-offset=(uint)27727, timestamp-offset=(uint)1713951204, ssrc=(uint)2573237941, a-framerate=(string)30" ! rtph265depay ! h265parse ! vaapidecode ! vaapipostproc ! vaapisink
Notice the h265parse element between the depay and decode
Related
I'm new for gstreamer. I want to encode the video of my MacbookPro. built-in cam to h264 and then play. in command line, I tried "
gst-launch-1.0 autovideosrc ! queue ! x264enc ! avdec_h264 ! queue ! autovideosink " and it works.but when I run the c++ code, it failed, only show a green screen.
video_src = gst_element_factory_make("autovideosrc", "video_source");
video_enc = gst_element_factory_make("x264enc", "videoEncoder");
video_dec = gst_element_factory_make("avdec_h264", "videodecoder");
video_sink = gst_element_factory_make("osxvideosink", nullptr);
gst_bin_add_many...
gst_element_link_many (video_src, screen_queue, video_enc, video_dec, video_sink, NULL);
not sure how to correct it. thanks!
Constructing this pipeline from code in few steps.
Gstreamermm used.
Create appsrc_1:
appsrc = Gst::ElementFactory::create_element("appsrc", name);
appsrc->set_property("block", false);
appsrc->set_property("min-latency", 0);
appsrc->set_property("max-latency", 100);
appsrc->set_property("do-timestamp", false);
appsrc->set_property("format", 3);
const char * format = nullptr;
switch (m_type)
{
case CV_8UC1: format = "GRAY8"; break;
case CV_8UC3: format = "BGR"; break;
default:
assert(false);
return;
}
appsrcCaps = Gst::Caps::create_simple("video/x-raw",
"format", format,
"framerate", Gst::Fraction(m_fps, 1),
"width", m_frameSize.width,
"height", m_frameSize.height);
appsrc->set_property("caps", appsrcCaps);
Create bin0:
bin = Glib::RefPtr<Gst::Bin>::cast_dynamic(Gst::Parse::create_bin("queue ! videoconvert", false));
binSinkPad = bin->add_ghost_pad(bin->get_element("queue0"), "sink");
binSrcPad = bin->add_ghost_pad(bin->get_element("videoconvert0"), "src");
Create tee:
tee = Gst::ElementFactory::create_element("tee");
tee->set_property("allow-not-linked", true);
Then, new pipeline was created and existing elements were added:
pipeline = Gst::Pipeline::create("webrtc_pipeline");
pipeline->add(appsrc);
pipeline->add(tee)->add(bin);
Link step:
appsrc->link(bin)->link(tee);
Then, on some time, when new client connects, bin1 spawns with fake ximagesink:
pipeline->set_state(Gst::STATE_READY);
bin1 = Glib::RefPtr<Gst::Bin>::cast_dynamic(Gst::Parse::create_bin("queue name=q ! ximagesink", false));
videoSinkPad = bin1->add_ghost_pad(bin1->get_element("q"), "sink", "videosinkpad");
bin1->set_state(Gst::STATE_PAUSED);
pipeline->add(bin1);
tee->link(bin1);
pipeline->set_state(Gst::STATE_PLAYING);
After those steps I am getting this strange log and error:
0:01:13.399425680 32822 0x7fd578002800 DEBUG videoconvert gstvideoconvert.c:405:gst_video_convert_transform_caps:<videoconvert0> transformed video/x-raw, format=(string)GRAY8, framerate=(fraction)8/1, width=(int)640, height=(int)480 into video/x-raw, format=(string){ I420, YV12, YUY2, UYVY, AYUV, VUYA, RGBx, BGRx, xRGB, xBGR, RGBA, BGRA, ARGB, ABGR, RGB, BGR, Y41B, Y42B, YVYU, Y444, v210, v216, Y210, Y410, NV12, NV21, GRAY8, GRAY16_BE, GRAY16_LE, v308, RGB16, BGR16, RGB15, BGR15, UYVP, A420, RGB8P, YUV9, YVU9, IYU1, ARGB64, AYUV64, r210, I420_10BE, I420_10LE, I422_10BE, I422_10LE, Y444_10BE, Y444_10LE, GBR, GBR_10BE, GBR_10LE, NV16, NV24, NV12_64Z32, A420_10BE, A420_10LE, A422_10BE, A422_10LE, A444_10BE, A444_10LE, NV61, P010_10BE, P010_10LE, IYU2, VYUY, GBRA, GBRA_10BE, GBRA_10LE, BGR10A2_LE, GBR_12BE, GBR_12LE, GBRA_12BE, GBRA_12LE, I420_12BE, I420_12LE, I422_12BE, I422_12LE, Y444_12BE, Y444_12LE, GRAY10_LE32, NV12_10LE32, NV16_10LE32, NV12_10LE40 }, width=(int)640, height=(int)480, framerate=(fraction)8/1
0:01:13.399443298 32822 0x7fd578002800 DEBUG videoconvert gstvideoconvert.c:405:gst_video_convert_transform_caps:<videoconvert0> transformed video/x-raw, format=(string)GRAY8, framerate=(fraction)8/1, width=(int)640, height=(int)480 into video/x-raw, framerate=(fraction)8/1, width=(int)640, height=(int)480
0:01:13.399507923 32822 0x7fd578002800 DEBUG videoconvert gstvideoconvert.c:342:gst_video_convert_fixate_caps:<videoconvert0> trying to fixate othercaps video/x-raw, framerate=(fraction)8/1, width=(int)640, height=(int)480, format=(string)BGRx, pixel-aspect-ratio=(fraction)5/3 based on caps video/x-raw, format=(string)GRAY8, framerate=(fraction)8/1, width=(int)640, height=(int)480
0:01:13.399515137 32822 0x7fd578002800 DEBUG videoconvert gstvideoconvert.c:353:gst_video_convert_fixate_caps:<videoconvert0> now fixating video/x-raw, framerate=(fraction)8/1, width=(int)640, height=(int)480, format=(string)BGRx, pixel-aspect-ratio=(fraction)5/3
0:01:13.399519528 32822 0x7fd578002800 DEBUG videoconvert gstvideoconvert.c:290:gst_video_convert_fixate_format:<videoconvert0> source format GRAY8
0:01:13.399524394 32822 0x7fd578002800 DEBUG videoconvert gstvideoconvert.c:300:gst_video_convert_fixate_format:<videoconvert0> iterate 1 structures
0:01:13.399528710 32822 0x7fd578002800 DEBUG videoconvert gstvideoconvert.c:264:score_value:<videoconvert0> score GRAY8 -> BGRx = 3
0:01:13.399532151 32822 0x7fd578002800 DEBUG videoconvert gstvideoconvert.c:269:score_value:<videoconvert0> found new best 3
0:01:13.399550504 32822 0x7fd578002800 ERROR videoconvert gstvideoconvert.c:490:gst_video_convert_set_info:<videoconvert0> input and output formats do not match
...
0:01:13.399741982 32822 0x7fd578002800 WARN basetransform gstbasetransform.c:1370:gst_base_transform_setcaps:<videoconvert0> FAILED to configure incaps video/x-raw, format=(string)GRAY8, framerate=(fraction)8/1, width=(int)640, height=(int)480 and outcaps video/x-raw, framerate=(fraction)8/1, width=(int)640, height=(int)480, format=(string)BGRx, pixel-aspect-ratio=(fraction)5/3
0:01:13.399756085 32822 0x7fd578002800 INFO task gsttask.c:312:gst_task_func:<queue0:src> Task going to paused
0:01:13.524512587 32822 0x7fd578002360 WARN basesrc gstbasesrc.c:3072:gst_base_src_loop:<appsrc_1> error: Internal data stream error.
0:01:13.524526933 32822 0x7fd578002360 WARN basesrc gstbasesrc.c:3072:gst_base_src_loop:<appsrc_1> error: streaming stopped, reason not-negotiated (-4)
The question is: How to fix this?
PS: How this could happen, if upstream is limited to format=GRAY8???
0:01:13.399507923 32822 0x7fd578002800 DEBUG videoconvert gstvideoconvert.c:342:gst_video_convert_fixate_caps:<videoconvert0> trying to fixate othercaps video/x-raw, framerate=(fraction)8/1, width=(int)640, height=(int)480, format=(string)BGRx, pixel-aspect-ratio=(fraction)5/3 based on caps video/x-raw, format=(string)GRAY8, framerate=(fraction)8/1, width=(int)640, height=(int)480
0:01:13.399515137 32822 0x7fd578002800 DEBUG videoconvert gstvideoconvert.c:353:gst_video_convert_fixate_caps:<videoconvert0> now fixating video/x-raw, framerate=(fraction)8/1, width=(int)640, height=(int)480, format=(string)BGRx, pixel-aspect-ratio=(fraction)5/3
I think your problem is not the pixel format=GRAY8 but the pixel-aspect-ratio=5/3 which is not the same up- and downstream of your videoconvert element.
videoconvert cannot handle conversion between different pixel-aspect-ratios. As can be seen here.
Check that you have the right resolution set for your ximagesink. ximagesink docs
I want to send raw audio data(PCM) to VLC player on RTP for playing the PCM using gstreamer.
Here is the command to send the PCM
gst-launch-1.0 -v filesrc location=/home/webos/pcm_data_dump ! audio/x-raw, rate=44100, channels=2, endianness=1234, format=S16LE, layout=interleaved, clock-rate=44100 ! audioconvert ! audioresample ! audio/x-raw, rate=44100, channels=2, format=S32LE, layout=interleaved ! audioconvert ! rtpL16pay pt=10 ! application/x-rtp, pt=10, encoding-name=L16, payload=10, clock-rate=44100, channels=2 ! udpsink host=192.168.0.2 port=5555
Here is the VLC option to receive the PCM
rtp://192.168.0.2:5555
VLC player can get the PCM from gstreamer, but it cannot play.
VLC shows the debug message like below.
Lastly "core debug: Buffering 0%" message is shown repeatedly in VLC debug message.
core debug: output 'f32l' 44100 Hz Stereo frame=1 samples/8 bytes
core debug: looking for audio volume module matching "any": 2 candidates
core debug: using audio volume module "float_mixer"
core debug: input 's16l' 44100 Hz Stereo frame=1 samples/4 bytes
core debug: looking for audio filter module matching "scaletempo": 14 candidates
scaletempo debug: format: 44100 rate, 2 nch, 4 bps, fl32
scaletempo debug: params: 30 stride, 0.200 overlap, 14 search
scaletempo debug: 1.000 scale, 1323.000 stride_in, 1323 stride_out, 1059
standing, 264 overlap, 617 search, 2204 queue, fl32 mode
core debug: using audio filter module "scaletempo"
core debug: conversion: 's16l'->'f32l' 44100 Hz->44100 Hz Stereo->Stereo
core debug: looking for audio converter module matching "any": 12 candidates
audio_format debug: s16l->f32l, bits per sample: 16->32
core debug: using audio converter module "audio_format"
core debug: conversion pipeline complete
core debug: conversion: 'f32l'->'f32l' 44100 Hz->44100 Hz Stereo->Stereo
core debug: Buffering 0%
core debug: conversion pipeline complete
core debug: looking for audio resampler module matching "any": 3 candidates
core debug: Buffering 0%
core debug: Buffering 0%
core debug: Buffering 0%
core debug: Buffering 0%
core debug: Buffering 0%
.......
And, the log below is shown once the gstreamer command to send PCM starts.
Normally, gstreamer is blocked with this message"New clock: GstSystemClock" when command starts.
Setting pipeline to PAUSED ...
Pipeline is PREROLLING ...
/GstPipeline:pipeline0/GstCapsFilter:capsfilter0.GstPad:src: caps = audio/x- raw, format=(string)S32LE, layout=(string)interleaved, rate=(int)44100, channels=(int)2
/GstPipeline:pipeline0/GstQueue:queue0.GstPad:sink: caps = audio/x-raw, format=(string)S32LE, layout=(string)interleaved, rate=(int)44100, channels= (int)2
/GstPipeline:pipeline0/GstQueue:queue0.GstPad:sink: caps = audio/x-raw, format=(string)S32LE, layout=(string)interleaved, rate=(int)44100, channels= (int)2
/GstPipeline:pipeline0/GstAudioConvert:audioconvert0.GstPad:src: caps = audio/x-raw, layout=(string)interleaved, rate=(int)44100, format=(string)S16BE, channels=(int)2, channel-mask=(bitmask)0x0000000000000003
/GstPipeline:pipeline0/GstQueue:queue1.GstPad:sink: caps = audio/x-raw, layout=(string)interleaved, rate=(int)44100, format=(string)S16BE, channels=(int)2, channel-mask=(bitmask)0x0000000000000003
/GstPipeline:pipeline0/GstQueue:queue1.GstPad:sink: caps = audio/x-raw, layout=(string)interleaved, rate=(int)44100, format=(string)S16BE, channels=(int)2, channel-mask=(bitmask)0x0000000000000003
/GstPipeline:pipeline0/GstRtpL16Pay:rtpl16pay0.GstPad:src: caps = application/x-rtp, media=(string)audio, clock-rate=(int)44100, encoding-name=(string)L16, encoding-params=(string)2, channels=(int)2, payload=(int)10, ssrc=(uint)2226113402, timestamp-offset=(uint)1744959080, seqnum-offset=(uint)62815
/GstPipeline:pipeline0/GstUDPSink:udpsink0.GstPad:sink: caps = application/x-rtp, media=(string)audio, clock-rate=(int)44100, encoding-name=(string)L16, encoding-params=(string)2, channels=(int)2, payload=(int)10, ssrc=(uint)2226113402, timestamp-offset=(uint)1744959080, seqnum-offset=(uint)62815
/GstPipeline:pipeline0/GstRtpL16Pay:rtpl16pay0.GstPad:sink: caps = audio/x-raw, layout=(string)interleaved, rate=(int)44100, format=(string)S16BE, channels=(int)2, channel-mask=(bitmask)0x0000000000000003
/GstPipeline:pipeline0/GstAudioConvert:audioconvert0.GstPad:sink: caps = audio/x-raw, format=(string)S32LE, layout=(string)interleaved, rate=(int)44100, channels=(int)2
/GstPipeline:pipeline0/GstRtpL16Pay:rtpl16pay0: timestamp = 1744959080
/GstPipeline:pipeline0/GstRtpL16Pay:rtpl16pay0: seqnum = 62815
Pipeline is PREROLLED ...
Setting pipeline to PLAYING ...
New clock: GstSystemClock
Got EOS from element "pipeline0".
Execution ended after 0:00:00.622147167
Setting pipeline to PAUSED ...
Setting pipeline to READY ...
Setting pipeline to NULL ...
Freeing pipeline ...
gst-launch-0.1 has no problem, only 1.0 has problem.
Is there any problem?
If I replace your filesrc with audiotestsrc, the example works for me. Still let me point out some room for improvement.
use audioparse instead of the first capsfilter
don't audioconvert twice.
Here is a simplified pipeline that works for me:
gst-launch-1.0 -v audiotestsrc ! audioresample ! audioconvert ! rtpL16pay pt=10 ! application/x-rtp, pt=10, encoding-name=L16, payload=10, clock-rate=44100, channels=2 ! udpsink host=localhost port=5555
I'd like to use pipeline below to play content with sound and without sound. Problem is that content without sound PREROLLING pipeline, but doesn't play
gst-launch-1.0.exe uridecodebin uri=file:///home/mymediafile.ogv name=d1 ! tee name=t1 ! queue max-size-buffers=2 ! jpegenc ! appsink name=myappsink t1. ! queue ! autovideosink d1. ! queue ! audioconvert ! audioresample ! autoaudiosink
How can I solve such issue?
I found no way to get your pipeline going on the command line. If I put in the audio portion of the pipeline, the files with no audio hang.
In your application however, you'll be able to add a signal for the pad_added events, and only added the audio portion of the pipeline when needed. Some pseudo code:
void decodebin_pad_added(GstElement *decodebin, GstPad *new_pad, gpointer user_data) {
GstElement* pipeline = (GstElement*)user_data;
GstCaps* audio_caps = gst_caps_from_string("audio/x-raw");
GstCaps* pad_caps = gst_pad_get_current_caps(new_pad);
if(! gst_caps_can_intersect(pad_caps, audio_caps)) {
return;
}
GstElement* audio_pipeline = gst_parse_launch("queue ! audioconvert ! audioresample ! autoaudiosink", NULL);
gst_bin_add(GST_BIN(pipeline), audio_pipeline);
GstElement* decodebin = gst_bin_get_by_name(GST_BIN(pipeline), "d1");
gst_element_link(decodebin, audio_pipeline);
gst_object_unref(decodebin);
}
void decodebin_no_more_pads(GstElement *decodebin, gpointer user_data) {
GstElement* pipeline = (GstElement*)user_data;
gst_element_set_state(pipeline, GST_PLAYING);
}
GstElement* pipeline = gst_parse_launch("uridecodebin uri=file:///home/mymediafile.ogv name=d1 ! tee name=t1 ! queue max-size-buffers=2 ! jpegenc ! appsink name=myappsink t1. ! queue ! autovideosink", NULL);
GstElement* decodebin = gst_bin_get_by_name(GST_BIN(pipeline), "d1");
g_signal_connect(decodebin, "pad-added", G_CALLBACK(decodebin_pad_added), pipeline);
g_signal_connect(decodebin, "no-more-pads", G_CALLBACK(decodebin_no_more_pads), pipeline);
gst_element_set_state(pipeline, GST_STATE_PAUSED); //pause to make demuxer and decoders get setup and find out what's in the file
Add async-handling=true to the autoaudiosink.
gst-launch-1.0.exe uridecodebin uri=file:///home/mymediafile.ogv
name=d1 ! tee name=t1 ! queue max-size-buffers=2 ! jpegenc ! appsink
name=myappsink t1. ! queue ! autovideosink d1. ! queue ! audioconvert
! audioresample ! autoaudiosink async-handling=true
I am new to GStreamer and I try to swap the color channels of a RGB-video. (e.g. red to blue). How can I do this with gst-launch?
I go trough this list but I am unable to find an element to do it: http://gstreamer.freedesktop.org/documentation/plugins.html
I wrote now my own Element. I used "Colorflip" as my base Element, changed the name to "ChannelFlip" (you must rename all methods from gst_video_flip_bla to gst_channel_flip_bla and rename the structs).
Then I was able to register my element with:
gst_element_register(NULL, "channelflip", GST_RANK_NONE, GST_TYPE_CHANNEL_FLIP);
Then I added my enums to GstChannelFlipMethod and my properties to _GstChannelFlip. Changed caps to "RGB" and added my Code to gst_channel_flip_packed_simple and called it in gst_channel_flip_transform_frame instead of videoflip->process (videoflip, out_frame, in_frame); with:
GST_OBJECT_LOCK (videoflip);
//videoflip->process (videoflip, out_frame, in_frame);
gst_channel_flip_packed_simple(videoflip, out_frame, in_frame);
GST_OBJECT_UNLOCK (videoflip);
You can actually trick GStreamer by replacing the caps:
gst-launch-1.0 -v videotestsrc ! video/x-raw, format=RGBx ! capssetter replace=true caps="video/x-raw, format=(string)BGRx, width=(int)320, height=(int)240, framerate=(fraction)30/1, multiview-mode=(string)mono, pixel-aspect-ratio=(fraction)1/1, interlace-mode=(string)progressive" ! videoconvert ! ximagesink
Please note that:
"width=(int)320, height=(int)240, framerate=(fraction)30/1, multiview-mode=(string)mono, pixel-aspect-ratio=(fraction)1/1, interlace-mode=(string)progressive"
are the default settings for videotestsrc. If you, for example, want another resolution, you need to declare it twice:
gst-launch-1.0 -v videotestsrc ! video/x-raw, format=RGBx, width=640, height=480 ! capssetter replace=true caps="video/x-raw, format=(string)BGRx, width=(int)640, height=(int)480, framerate=(fraction)30/1, multiview-mode=(string)mono, pixel-aspect-ratio=(fraction)1/1, interlace-mode=(string)progressive" ! videoconvert ! ximagesink
But of course having a dedicated element is the better solution in order to support proper dynamic caps negotiation.