I want to capture the RTSP stream from some IP cameras, and after looking around I found 2 great tools to do this: avconv and openRTSP
openRTSP -u user password rtsp://10.48.34.125/axis-media/media.amp
avconv -i "rtsp://user:password#10.48.34.125/axis-media/media.amp" -vcodec copy -f mp4 10.48.34.125.mp4
but for some voodoo reason when I need to use URLs without an specific extension, such as:
rtsp://user:password#10.48.34.46/
avconv returns 401 Unauthorized
so I'm stuck with openRTSP at the moment...
The thing is, unlike avconv, openRTSP outputs a raw file which is encoded to 25fps, which made some of my videos look like they where in fast-forward. I found a (cpu expensive) way to re-encode the file to a closer frame rate to what I need:
avconv -r 7 -i video-H264-1 -r 24 -f mp4 10.48.34.28.mp4
(in this example I'm forcing the frame rate of the raw file to be 7, and the frame rate of the output file to be 24. I tried using openRTSP build-in flags, but the output file still had a frame rate of 25: openRTSP -f 7 -u user password rtsp://10.48.34.145/mpeg4/media.3gp)
Sadly the video looks odd at certain points, and that's because the original stream sometimes has a variable frame rate (for example at night).
My question is, is there some way to deactive this default encondig to 25fps?
And why 25? I mean, isn't the norm 24?
try:
avconv -rtsp_transport tcp -i rtsp://server -an -vcodec copy -f mp4 10.48.34.28.mp4
if you want to change original video rate to 24 you must transcode it:
avconv -rtsp_transport tcp -i rtsp://server -an -vcodec libx264 -r 24 -f mp4 10.48.34.28.mp4
I have tried this example in order to segment a given video file using ffmpeg into an m3u8 file and smaller chunks (.ts files). This actually worked great. Is it possible to do practically the same thing with audio input?
This was my most promising approach so far (capturing live audio on Windows OS):
ffmpeg -f dshow -i audio="<name of input device>" -acodec libmp3lame -ab 64000 | segmenter - 10 stream stream.m3u8 http://<IP_OF_SERVER>/stream/stream/ 5 1
But this returns this error:
At least one output file must be specified.
Could not open input file, make sure it is an mpegts file: -1
I really would not know how to convert the live audio stream to an mpegts file.
Could anyone please give me a hint?
Thanks a lot
I am writing a small mp3 conversion tool. We upload a mp3 file & would like to convert it to a 96kbps file & a 320 kbps file. I have written the conversion script & it runs. But these files do not play.
Am i missing something?
the code i've written is:
/usr/local/bin/ffmpeg -i test.mp3 -vn -ar 441000 -ac 2 -ab 96k -f mp2 music/96/test.mp3 2>&1
Thanks!
There are two major issues here:
The audio rate you're setting (-ar 441000) is incorrect. You want 44.1 kHz (-ar 44100).
You're forcing the use of MPEG2 audio (-f mp2), which is not what you want, and is probably not supported by the player you're using either. Leave that option out entirely; the .mp3 extension on the output file will be used as a hint anyway.
I have a file, sound.au, which file describes as Sun/NeXT audo data: 8-bit ISDN mu-law, mono, 8000 Hz. I'd like to convert this to a WAV that file would describe as RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz. However, I cannot get the right set of arguments to make this conversion and see what it sounds like.
Has anyone performed this conversion or similar before? sox -t auto -w -s -r 8000 -c sound.au sound.wav gets me close, but it's G711 mu-law, not 16 bit PCM.
Thanks.
I don't have an .au file to try, but I suspect sox sound.au -e signed-integer sound.wav would work. You are only trying to change the encoding from u-law to PCM, right? sox should pick up all the necessary input info from the .au header. If it doesn't, maybe you need sox -t auto sound.au -e signed-integer sound.wav.
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.