I would like to store a file which has AAC audio frames,
For that i used the below pipeline,
gst-launch-1.0 filesrc location=Test_44100Hz_2ch_s16le.wav ! "audio/x-raw,rate=44100,format=s16le,channels=2" ! audioparse format=raw raw-format=s16le rate=44100 channels=2 ! faac ! aacparse ! queue ! filesink location=a1
While reading that file again to pulsesink using below pipeline,
gst-launch-1.0 filesrc location=a1 ! aacparse ! faad ! audioconvert ! audioresample ! pulsesink
I am Receiving below error, I used GST_DEBUG=3, but i am not able find the solution.
0:00:00.031924804 3379 0x2231d60 WARN basesrc gstbasesrc.c:3483:gst_base_src_start_complete:<filesrc0> pad not activated yet
Pipeline is PREROLLING ...
0:00:00.033044700 3379 0x2231050 WARN baseparse gstbaseparse.c:3255:gst_base_parse_loop:<aacparse0> error: No valid frames found before end of stream
ERROR: from element /GstPipeline:pipeline0/GstAacParse:aacparse0: No valid frames found before end of stream
Additional debug info:
gstbaseparse.c(3255): gst_base_parse_loop (): /GstPipeline:pipeline0/GstAacParse:aacparse0
ERROR: pipeline doesn't want to preroll.
Can anybody help me, To solve this? I need to store AAC audio frames and need to stream that file as AAC audio stream.
This is it, tested working:
gst-launch-1.0 filesrc location=WAV_44_16bit.wav ! decodebin ! audioconvert ! queue ! voaacenc ! aacparse ! queue ! mp4mux ! filesink location=aac.mp4
gst-launch-1.0 filesrc location=aac.mp4 ! decodebin ! audioconvert ! audioresample ! alsasink
In container there are metadata information stored.. without them the decoder does not know how to process the data.
AAC Audio streams require a container in order to be useful within gstreamer
For decoder initialization it is necessary to know sampling frequency and Audio Object. In gstreamer we are unable to pass this metadata directly to the parser or the decoder. The parser collects this data instead from the mp4 header then the encoder inherits the frame structure/size and sample rate. So this is a deficiency in either aacparse(parser) or avdec_aac/faad(decoder), none of which have exposed parameters to specify frame size of a raw file, the afore mentioned metadata. That being said, I haven't found a compelling reason why anyone would need to do this. I found myself trying to do it before I discovered the aac simply needed to be muxed into an MP4(mp4mux) or another container to work and be portable. The container/framing only adds a small amount of data to the stream.
Related
I'm having an issue with pulling audio and video from an RTSP stream using gstreamer.
The command I am using to test is as follows:
gst-launch-1.0 rtspsrc location=rtsp://192.168.50.160/whp name=src src. ! queue ! rtph264depay ! h264parse ! avdec_h264 ! videoconvert ! x264enc bitrate=10000 ! rtph264pay ! udpsink host=192.168.50.164 port=8004 src. ! queue ! fakesink
The result of the above is that the pipe follows through for the first (video) stream. The second stream however is untouched and seems to sit in the rtspsrc plugin.
The way I am finding this is by looking at the resultant dot file:
If I am reading this right it looks like the queue connects correctly to rtpsession0, but seems to ignore rtpsession1 and the second queue doesn't connect to anything resulting in audio from my stream being completely ignored.
Am I reading this incorrectly? If not am I missing something in my pipeline command that would rectify this issue?
I am happy to provide any more information necessary
Thanks
Recorded files with gstreamer-0.10 with FPS25 and FourCIF_Format plays in fast forward mode. Any solution would be appreciated. Some times skips 3-4 seconds in recorded files.
The pipeline I'm attempting to use is:
gst-launch v4l2src device=/dev/video2 !
'video/x-raw-yuv,width=704,height=576, framerate=25/1' ! tee
name=liveTee ! queue ! mfw_isink liveTee. ! queue ! vpuenc ! avimux !
filesink location=/home/Recording.avi
I'm gonna take a rough stab at it and re-format your question a bit. This is mostly a GStreamer and Freescale question, not so much QT.
gst-launch-1.0 -e videotestsrc pattern=ball do-timestamp=true
is-live=true ! timeoverlay !
'video/x-raw,width=704,height=576,framerate=25/1' ! tee name=liveTee !
queue leaky=downstream ! videoconvert !
ximagesink async=false
liveTee. ! queue leaky=downstream ! videoconvert ! queue ! x264enc !
avimux ! filesink location=/tmp/test.avi
The thing to keep in mind is that your encoder has to keep up with the live playback. So your pipeline needs to handle the case where the encoder falls out of sync. On the queue elements behind the tee, use the leaky attribute.
Then you also want to be careful about your video source and what it's supplying. It looks like in your case you want live video, but if your source was an existing video file the pipeline would probably need some more tweaking.
NOTE: It may be even simpler than that, just adding async=false to the videosink appears to be very important.
I'm trying to record stream from m3u8 file.
This pipeline works:
gst-launch-0.10 -e souphttpsrc location=(mysrc.m3u8) ! queue ! hlsdemux ! queue ! mpegtsparse ! queue ! mpegtsdemux ! queue ! audio/mpeg ! queue ! filesink location=test.ts
and (sometimes) record audio stream.
But i can't record video, whatever i do it crashes.
I tried something like this:
gst-launch-0.10 souphttpsrc location=(mysrc.m3u8) ! queue ! hlsdemux ! queue ! mpegtsparse ! queue ! mpegtsdemux ! queue ! video/x-264 ! queue ! filesink location=test.ts
But it does nothing.
You are using gstreamer 0.10 which is obsolete and unmantained, all users should upgrade to the 1.x series.
Given that warning, it is not clear whether you want to save the mpegts stream or the streams inside it.
To save the mpegts stream you can just do:
gst-launch-1.0 http://path/to/your/stream.m3u8 ! hlsdemux ! filesink
Be aware that if the HLS playlist contains multiple bitrates hlsdemux might switch bitrate and it will fail as gst-launch-1.0 isn't capable of handling this. (it is a debugging and testing tool). You can likely set a fixed "connection-speed" to make it always use the same bitrate you desire to overcome this issue.
If you want to get only the video stream and you know it is H264, try:
gst-launch-1.0 http://path/to/your/stream.m3u8 ! hlsdemux ! tsdemux ! queue ! video/x-h264 ! filesink
It might be a better idea to save it to a container format to allow easier use later, with something like:
gst-launch-1.0 http://path/to/your/stream.m3u8 ! hlsdemux ! tsdemux ! queue ! video/x-h264 ! h264parse ! qtmux ! filesink
But, as I said, please move to 1.x, HLS is much better at 1.x than it was in 0.10 and it should work.
I used the following GStreamer pipeline to store my encoded stream in a binary file:
gst-launch v4l2src ! videorate ! video/x-raw-yuv, framerate=\(fraction\)10/1 \
! videoscale ! video/x-raw-yuv, format=\(fourcc\)I420, width=640, height=480\
! ffmpegcolorspace ! x264enc ! fdsink > vid.bin
Now i want to play previously recorded files in GStreamer using the following pipeline:
cat vid.bin | gst-launch fdsrc ! ffdec_h264 ! autovideosink
But then it gives the following error:
Setting pipeline to PAUSED ...
Pipeline is PREROLLING ...
ERROR: from element /GstPipeline:pipeline0/ffdec_h264:ffdec_h2640: Internal GStreamer error: negotiation problem. Please file a bug at http://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer.
Additional debug info:
gstffmpegdec.c(2804): gst_ffmpegdec_chain (): /GstPipeline:pipeline0/ffdec_h264:ffdec_h2640:
ffdec_h264: input format was not set before data start
ERROR: pipeline doesn't want to preroll.
Setting pipeline to NULL ...
Freeing pipeline ...
I know that the best way to capture video is using Muxers but is there any way to play my previous files?
Thanks
Not sure your pipeline is right.
If you want to write to a file why not simply use filesink and filesrc.
fdsink > vid.bin will not work fine because if you see the prints by gstreamer gst-launch will also go into the file. [Just open vid.bin in an text editor and you will see what I mean].
Also for x264 stream to be stored without a muxer you need to use byte-stream=1 in your x264enc to store it in annexb format so that it is decodable.
To play back raw x264 stream you need to have a color space convertor before the video sink
gst-launch filesrc location=inputfile ! legacyh264parse ! ffdec_h264 ! queue ! ffmpegcolorspace ! autovideosink
plays just fine here at my end
Or, to playback a raw h264 file with gstreamer 1.0:
gst-launch-1.0 filesrc location=/tmp/video.h264 ! h264parse ! avdec_h264 ! autovideosink
I am trying to create a GStreamer pipeline (v 1.0) in order to record and play special file format.
For recording purpose I use the following pipeline:
gst-launch-1.0 videotestsrc ! video/x-raw-yuv, format=\(fourcc\)I420, width=640, height=480 ! videoconvert ! x264enc byte-stream=1 ! queue ! appsink
In appsink (using new_sample() callback) I use a compression method to compress H264 stream and finally store in a output file.
I use the following pipeline to play the recorded file:
gst-launch-1.0 appsrc ! video/x-h264 ! avdec_h264 ! autovideosink
In appsrc I decompress H264 stream and send it to appsrc buffer (using push-buffer). The size of each buffer is 4095.
Unfortunately GStreamer after push 2 buffers print the following debug message:
Error: Internal data flow error.
Is there any way to fix the problem?
Add legacyh264parse or h264parse (depending on your version of gst components) before your decoder. You need to be able to send full frames to the decoder.
Post avdec_h264 it would be nice to have a ffmpegcolorspace to be able to convert the video format to your display requirements.