I am trying to record what my device is currently playing using arecord (it is playing audio via mplayer).
I've done everything I have been able to read so far such as checking under which user mplayer is running, setting the user as per the command below, but it only ever records silence.
This is the list of users where my user is ID 1000:
user:x:1000:1000:,,,:/home/user:/bin/bash
This shows that what I am trying to record is running by that user:
user 1217 2.3 1.3 131704 25524 ? SL 09:11 12:15 /usr/bin/mplayer http://example.com/stream
This is the command I am running:
XDG_RUNTIME_DIR=/run/user/1000 /usr/bin/arecord -d 1 /home/user/rec.waw
The command will output:
Recording WAVE '/home/user/rec.waw' : Unsigned 8 bit, Rate 8000 Hz, Mono
And it writes the file, but the content is silent.
I have tried specifying --device but not sure how to specify it from the list of devices below (I tried --device=CARD=Headphones and --device=hw:CARD=Headphones but I get errors.
This is the output of arecord -L
user#device:~ $arecord -L
null
Discard all samples (playback) or generate zero samples (capture)
jack
JACK Audio Connection Kit
pulse
PulseAudio Sound Server
default
Playback/recording through the PulseAudio sound server
output
usbstream:CARD=b1
bcm2835 HDMI 1
USB Stream Output
usbstream:CARD=Headphones
bcm2835 Headphones
USB Stream Output
Any help would be appreciated.
essentially I'm trying to just run mpv on a video, and after you quit you get some information about the video.
Example:
(+) Video --vid=1 (*) (hevc 1920x1080 60.000fps)
(+) Audio --aid=1 (*) (aac 2ch 48000Hz)
AO: [pulse] 48000Hz stereo 2ch float
VO: [gpu] 1920x1080 yuv420p
AV: 00:00:01 / 00:12:32 (0%) A-V: 0.000
Exiting... (Quit)
I would like to grep out the AV line, however this line is added before/afterwards (I'm not exactly sure). When running mpv video_file | grep AV
The video would play and the terminal would give this little status line
(Paused) AV: 00:00:02 / 00:00:57 (5%) A-V: 0.000 (random example video, different to the first one)
However after closing mpv there is no output, and its further confirmed when running
mpv video_file >> test.txt
giving an output of
(+) Video --vid=1 (*) (hevc 1920x1080 60.000fps)
(+) Audio --aid=1 (*) (aac 2ch 48000Hz)
AO: [pulse] 48000Hz stereo 2ch float
VO: [gpu] 1920x1080 yuv420p
Exiting... (Quit)
suggesting that it places the AV line outside of the output, between VO line and the space line.
Soooo like how would I grep out this line?
I am trying to stream video and audio from a Camera in a browser using Webrtc and Wowza Media Server (4.7.3 version).
The camera stream (h264/aac) is first of all transcoded by using FFMPEG (version N-89681-g2477bfe built with gcc 4.8.5, last available version on ffmpeg website) in VP8/OPUS and then pushed to the Wowza Server.
By using the small Wowza webpage I ask for the Wowza stream to be displayed in the browser (Chrome Version 66.0.3336.5 Build officiel canary 32 bits).
FFMPEG used command :
ffmpeg -rtsp_transport tcp -i rtsp://<camera_stream> -vcodec libvpx -vb 600000 -crf 10 -qmin 0 -qmax 50 -acodec libopus -ab 32000 -ar 48000 -ac 2 -f rtsp rtsp://<IP_Address_Wowza>:<port_no_ssl>/<application_name>/test
When I click on Play stream I have a very bad quality video and audio (jerky video and very bad audio).
If I use this FFMPEG command:
ffmpeg -rtsp_transport tcp -i rtsp://<camera_stream> -vcodec libvpx -vb 600000 -crf 10 -qmin 0 -qmax 50 -acodec copy -f rtsp rtsp://<IP_Address_Wowza>:<port_no_ssl>/<application_name>/test
I will have a good video (flowing, smooth) but no audio (the camera micro is ON).
If libopus is the problem (as this test first shows), I tried libvorbis but with Chrome console I have this error "Failed to set remote offer sdp: Session error code: ERROR_CONTENT". Weird, cause libvorbis is one of the available codecs for Webrtc.
Is someone experiencing the same issue ? Did someone experience the same issue ?
Thanks in advance.
You probably have no audio because opus must have sample rate of 48000
You should add the flag:
"-ar 48000"
to the output settings
I also experienced the "bad quality video and audio issues".
I finally solved the issue by adding:
"-quality realtime" to the output settings .
That work well for me, I hope this will help you.
I want to convert a MOV from my Casio cam to mp4 using transcode. Why transcode? Because I also want to deshake the video in the same step.
When I use
ffmpeg -i in.MOV out.mp4
it works. When using
transcode -J stabilize -i in.MOV
or
transcode -J transform -i in.MOV -y ffmpeg -F mpeg4 -o out.mp4
I get hundreds of these errors:
[ffmpeg_audio] Error: avcodec_open2 failed
[adpcm_ima_wav # 0x1f7f180] Only 4-bit ADPCM IMA WAV files are supported
This looks to me as if transcode uses ffmpeg internally.
I could use ffmpeg to make it mp4 first and then use transcode to stabilize the video, but then it would be re-encoded twice which I would like to avoid.
This is what mplayer says about my MOV file:
MPlayer2 2.0-701-gd4c5b7f-2ubuntu2 (C) 2000-2012 MPlayer Team
Cannot open file '/home/koem/.mplayer/input.conf': No such file or directory
Failed to open /home/koem/.mplayer/input.conf.
Cannot open file '/etc/mplayer/input.conf': No such file or directory
Failed to open /etc/mplayer/input.conf.
Playing 1-original.MOV.
Detected file format: QuickTime / MOV (libavformat)
[lavf] stream 0: video (h264), -vid 0
[lavf] stream 1: audio (adpcm_ima_wav), -aid 0, -alang eng
Clip info:
major_brand: qt
minor_version: 537921536
compatible_brands: qt caqv
creation_time: 2017-01-02 23:31:38
Load subtitles in .
Failed to open VDPAU backend libvdpau_i965.so: cannot open shared object file: No such file or directory
[vdpau] Error when calling vdp_device_create_x11: 1
[ass] auto-open
Selected video codec: H.264 / AVC / MPEG-4 AVC / MPEG-4 part 10 [libavcodec]
Selected audio codec: ADPCM IMA WAV [libavcodec]
AUDIO: 44100 Hz, 2 ch, s16le, 352.8 kbit/25.00% (ratio: 44100->176400)
AO: [pulse] 44100Hz 2ch s16le (2 bytes per sample)
Starting playback...
VIDEO: 1920x1080 29.970 fps 15940.0 kbps (1992.5 kB/s)
VO: [xv] 1920x1080 => 1920x1080 Planar YV12
Colorspace details not fully supported by selected vo.
A: 1.1 V: 1.1 A-V: -0.000 ct: 0.000 0/ 0 16% 8% 1.6% 0 0
Exiting... (Quit)
How can I make it work with transcode without using ffmpeg first?
FFmpeg has a deshake as well as a stabilization filter. Get a new binary if yours doesn't.
To continue with your existing binaries, run
ffmpeg -i in.MOV -vcodec copy out.mp4
This will skip video re-encoding.
I want my website to join some webcam recordings in FLV files (like this one). This needs to be done on Linux without user input. How do I do this? For simplicity's sake, I'll use the same flv as both inputs in hope of getting a flv that plays the same thing twice in a row.
That should be easy enough, right? There's even a full code example in the ffmpeg FAQ.
Well, pipes seem to be giving me problems (both on my mac running Leopard and on Ubuntu 8.04) so let's keep it simple and use normal files. Also, if I don't specify a rate of 15 fps, the visual part plays extremely fast. The example script thus becomes:
ffmpeg -i input.flv -vn -f u16le -acodec pcm_s16le -ac 2 -ar 44100 \
- > temp.a < /dev/null
ffmpeg -i input.flv -an -f yuv4mpegpipe - > temp.v < /dev/null
cat temp.v temp.v > all.v
cat temp.a temp.a > all.a
ffmpeg -f u16le -acodec pcm_s16le -ac 2 -ar 44100 -i all.a \
-f yuv4mpegpipe -i all.v -sameq -y output.flv
Well, using this will work for the audio, but I only get the video the first time around. This seems to be the case for any flv I throw as input.flv, including the movie teasers that come with red5.
a) Why doesn't the example script work as advertised, in particular why do I not get all the video I'm expecting?
b) Why do I have to specify a framerate while Wimpy player can play the flv at the right speed?
The only way I found to join two flvs was to use mencoder. Problem is, mencoder doesn't seem to join flvs:
mencoder input.flv input.flv -o output.flv -of lavf -oac copy \
-ovc lavc -lavcopts vcodec=flv
I get a Floating point exception...
MEncoder 1.0rc2-4.0.1 (C) 2000-2007 MPlayer Team
CPU: Intel(R) Xeon(R) CPU 5150 # 2.66GHz (Family: 6, Model: 15, Stepping: 6)
CPUflags: Type: 6 MMX: 1 MMX2: 1 3DNow: 0 3DNow2: 0 SSE: 1 SSE2: 1
Compiled for x86 CPU with extensions: MMX MMX2 SSE SSE2
success: format: 0 data: 0x0 - 0x45b2f
libavformat file format detected.
[flv # 0x697160]Unsupported audio codec (6)
[flv # 0x697160]Could not find codec parameters (Audio: 0x0006, 22050 Hz, mono)
[lavf] Video stream found, -vid 0
[lavf] Audio stream found, -aid 1
VIDEO: [FLV1] 240x180 0bpp 1000.000 fps 0.0 kbps ( 0.0 kbyte/s)
[V] filefmt:44 fourcc:0x31564C46 size:240x180 fps:1000.00 ftime:=0.0010
** MUXER_LAVF *****************************************************************
REMEMBER: MEncoder's libavformat muxing is presently broken and can generate
INCORRECT files in the presence of B frames. Moreover, due to bugs MPlayer
will play these INCORRECT files as if nothing were wrong!
*******************************************************************************
OK, exit
Opening video filter: [expand osd=1]
Expand: -1 x -1, -1 ; -1, osd: 1, aspect: 0.000000, round: 1
==========================================================================
Opening video decoder: [ffmpeg] FFmpeg's libavcodec codec family
Selected video codec: [ffflv] vfm: ffmpeg (FFmpeg Flash video)
==========================================================================
audiocodec: framecopy (format=6 chans=1 rate=22050 bits=16 B/s=0 sample-0)
VDec: vo config request - 240 x 180 (preferred colorspace: Planar YV12)
VDec: using Planar YV12 as output csp (no 0)
Movie-Aspect is undefined - no prescaling applied.
videocodec: libavcodec (240x180 fourcc=31564c46 [FLV1])
VIDEO CODEC ID: 22
AUDIO CODEC ID: 10007, TAG: 0
Writing header...
[NULL # 0x67d110]codec not compatible with flv
Floating point exception
c) Is there a way for mencoder to decode and encode flvs correctly?
So the only way I've found so far to join flvs, is to use ffmpeg to go back and forth between flv and avi, and use mencoder to join the avis:
ffmpeg -i input.flv -vcodec rawvideo -acodec pcm_s16le -r 15 file.avi
mencoder -o output.avi -oac copy -ovc copy -noskip file.avi file.avi
ffmpeg -i output.avi output.flv
d) There must be a better way to achieve this... Which one?
e) Because of the problem of the framerate, though, only flvs with constant framerate (like the one I recorded through facebook) will be converted correctly to avis, but this won't work for the flvs I seem to be recording (like this one or this one). Is there a way to do this for these flvs too?
Any help would be very appreciated.
I thought it would be a nice learning exercise to rewrite it in Ruby.
It was.
Six months later and three gems later, here's the released product.
I'll still be working a bit on it, but it works.
You'll encounter a very subtle problem here because most video and audio formats (especially in ordinary containers) use "global headers," meaning at the start of the file they have a single header which specifies compression information (like width, height, etc) for the whole file. Concatting two streams will clearly fail, as it will now have two headers instead of one and the muxer may not like this. Converting to AVI probably is resolving the issue in your case because mencoder has code to concat AVIs--that code properly handles the header issue.
After posting my question on mencoder's mailing list, trying other things, I resorted to write my own tool! I started from flvtool and after some digging in the code and writing about 40 lines of code, it works, with no loss in quality (since there is no transcoding).
I'll release it asap, in the meantime anyone interested can contact me.
dont know if this will actually work but try using this command :
cat yourVideos/*.flv >> big.flv
this will probably damage meta information so after executing that command use "flvtool" (ruby script you can find it with google) to fix it.