My team accidentally on purpose clicked NO when Audacity asked to save the recording. So I left with bunch of *.au files, after using recovery program.
Some of them did have header and still open-able with audacity itself (example : this one), and some other are just complete nonsense, sometimes having the header filled with text from random javascript or HTML code (like this one). Probably hard disk half overwritten with browser cache? I don't know. And at this point, I almost don't care.
The audacity is on default settings, with sample rate 44100Hz. I can open a-113.au using audacity, from standard open files. I also was able to achieve open files using "Open RAW files" from Audacity, using this settings :
so I assume header takes 12384 bytes.
Now, how do I trim 12384 bytes from the file when opened as RAW, with either SoX or ffmpeg? because if I open it as RAW with 0 offset (default settings), it will add the header as a noise.
Current ffmpeg command I used : ffmpeg.exe -f f32le -ar 44.1k -ac 1 -i source destination
Current sox command I used : sox -t raw --endian little --rate 44100 -b 1 -b 32 --encoding floating-point %%A "converted/%%~nxA.wav"
Both still have header as a noise in the converted files.
With ffmpeg, use the subfile protocol.
ffmpeg.exe -f f32le -ar 44.1k -ac 1 -start 12384 -i "subfile:source" destination
I'm using latest static build of ffmpeg windows.
My input file (.mkv) is:
[video] - 1080, V_MPEG4/ISO/AVC, 14.6 Mbps, ID#0
[audio] - DTS 5.1, 1510 Kbps, ID#1
[subtitles] - S_TEXT/ASS Lossless English, ID#14
My problem is this: I convert the audio, so that my target player, a XB1 console (media support faq), is able to play audio/video. However sometimes its rather difficult to hear or parts may be in foreign language, so I want to force the english subtitles into the mix at the same time I convert the audio.
Currently for the audio, I use the following command
ffmpeg -i input.mkv -codec copy -acodec ac3 output.mkv
Can I somehow tie in the forced subtitles (onto the video) in order to save an extra process of taking the output.mkv and trying to force subtitles on?
Edit: I've tried using the following command to extract subtitles to be able to edit them
ffmpeg -i Movie.mkv -map 0:s:14 subs.srt
However i get the error: Stream map '0:s:14' matches no streams
Edit2: attempted to extract subtitles and succeeded with
ffmpeg -i input.mkv -map 0:14 -c copy subtitles.ass
but still looking to force the subtitles, nonetheless!
Also - a little bonus to this question - can I somehow extract the .ass file and edit it to only produce subtitles for foreign parts - so english audio doesn't have subtitles during the movie but foreign audio does have subtitles?
Cheers
Edit3:
When I try to use both of the commands at once (my earlier mentioned audio converter & one from the ffmpeg wiki)
ffmpeg -i input.mkv -codec copy -acodec ac3 -vf "ass=subs.ass" output.mkv
I get the following error from ffmpeg,
Filtergraph 'ass=subs.ass' was defined for video output stream 0:0 but codec copy was selected.
Filtering and streamcopy cannot be used together.
Since your media player does not support subtitles, the text has to be burnt onto the video image. For that, use
ffmpeg -i input.mkv -vf "ass=subs.ass" -c:v libx264 -crf 20 -c:a ac3 output.mkv
This will re-encode the video, since text is being added. The CRF value controls the video quality. Lower values produce better quality but larger files. 18 to 28 is a decent range to try.
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 have been using FFMPEG for some time for audio file conversions. I just downloaded the newest version and the command below:
ffmpeg -i x.flac -ab 128k x.mp3
no longer produces a 128k file.
It produces a 48k file no matter what the -ab command requests. Any suggestions?
According to the documentation
`-ab bitrate'
Set the audio bitrate in bit/s (default = 64k).
So this is the bitrate, which means that per second by default 64k bits are used for storing sound information. So only a one second file would return a 128k file.
If I misunderstood your question and you mean that the bitrate is 48k instead of 128k then double check if ffmpeg is not using variable bit rate. If this is not the case I suggest you submit a bug report.
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.