I have a mov file :
Metadata:
timecode: 09:59:50:00
Duration: 00:00:30.00, bitrate: 117714 kb/s
Stream #0.0(eng): Video: dvvideo, yuv422p, 1440x1080i tff [PAR 4:3 DAR 16:9]
, 115200 kb/s, 25.00 fps
Metadata:
codec_name: DVCPRO HD 1080i50
Stream #0.1(eng): Audio: pcm_s16le, 48000 Hz, 2 channels, s16, 1536 kb/s
Stream #0.2(eng): Data: unknown (tmcd)
I can see from MediaInfo
That the Audio is Muxed into the video. I'm trying to re-wrap this into an XDCAM, and copy over the audio streams. The problem is that I don't know how to map the audio that is wrapped into the video?
This is the command I have so far:
ffmbc -threads 8 -i "input.mov" -threads 8 -tff
-pix_fmt yuv422p -vcodec mpeg2video -timecode 09:59:50:00
.. other tags omitted ..
-acodec pcm_s24le
-map_audio_channel 0.1:0-0.1:0
-map_audio_channel 0.1:1-0.1:1
-f mov -y "output.mov"
-acodec pcm_s24le
-map_audio_channel 0.2:0-0.2:0
-map_audio_channel 0.2:1-0.2:1 -newaudio
When executed this returns "Cannot find audio channel 0.2.0". I changed the input stream identifier to stream 0, and 1 for the audios. Which when executed returned "Cannot find audio channel #0.0.0" presumably because it's trying to find a audio channel within the video stream?
How can I extract the audio from this file?
You may notice I'm using FFMBC, not FFMPEG ( there is no tag for FFMBC ), but I imagine it's the same for both. I'm not constrained to FFMBC, I can move to FFMPEG if it has a solution.
Thanks
Related
I've been using this command to convert my public rtmp audio/video stream to a local mp3 audio icecast2 stream, but I have been unable to do the same for both video and audio.
[Audio Only] (This works fine)
ffmpeg -re -i rtmp://162.142.xx.xxx:xxx/stream -vn -codec:a libmp3lame -b:a 128k -f mp3 -content_type audio/mpeg icecast://source:password#192.168.1.xxx:80/live
I've tried to re-write in order to support video, but I keep hitting dead ends
[Audio & Video Attempt] (this does not work)
ffmpeg -re -i rtmp://162.142.xx.xxx:xxx/stream -codec:v -f mpeg4 -b:v -f mpeg4 -content_type video/mpeg4 icecast://source:password#192.168.1.xxx:80/live
When I run this command, it gives me the error below asking for a suitable format.
$ ffmpeg -re -i rtmp://162.142.xx.xxx:xxx/stream -codec:v -f mpeg4 -b:v -f mpeg4 -content_type video/mpeg4 icecast://source:password#192.168.1.xxx:80/live
[h264 # 0x5598ffbb8980] co located POCs unavailable
[h264 # 0x5598ffbb8980] mmco: unref short failure
Input #0, flv, from 'rtmp://162.142.xx.xxx:xxx/stream':
Metadata:
|RtmpSampleAccess: true
Server : NGINX RTMP (github.com/sergey-dryabzhinsky/nginx-rtmp-module)
displayWidth : 1280
displayHeight : 720
fps : 48
videokeyframe_frequency: 0
profile :
level :
Duration: 00:00:00.00, start: 28117.779000, bitrate: N/A
Stream #0:0: Audio: aac (LC), 48000 Hz, stereo, fltp, 327 kb/s
Stream #0:1: Video: h264 (High), yuv420p(tv, bt709, progressive), 1280x720 [SAR 1:1 DAR 16:9], 2560 kb/s, 48 fps, 48 tbr, 1k tbn
[NULL # 0x5598ffb8bec0] Unable to find a suitable output format for 'mpeg4'
mpeg4: Invalid argument
I am positive that icecast2 can support video streams, however on the few occasions that I was able to actively stream successfully to it, it only showed an empty video embed.
I've re-written the command for AV multiple times while referencing ffmpeg documentation, however my above attempt seems to be the closest (concept-wise) that I have gotten.
What flags/formatting might I be missing which are causing the stream not to work?
I have some mpg files that I transcoded from DVDs I bought a long time ago (maybe 20 years ago). ffprobe:
Input #0, mpeg, from 'da-orig.mpg':
Duration: 00:06:59.44, start: 0.044100, bitrate: 6354 kb/s
Stream #0:0[0x1e0]: Video: mpeg2video (Main), yuv420p(tv, progressive), 720x480 [SAR 8:9 DAR 4:3], Closed Captions, 31 fps, 59.94 tbr, 90k tbn
Side data:
cpb: bitrate max/min/avg: 7500000/0/0 buffer size: 1835008 vbv_delay: N/A
Stream #0:1[0x85]: Audio: ac3, 48000 Hz, stereo, fltp, 192 kb/s
Stream #0:2[0x83]: Audio: ac3, 48000 Hz, stereo, fltp, 192 kb/s
Stream #0:3[0x81]: Audio: ac3, 48000 Hz, mono, fltp, 192 kb/s
Stream #0:4[0x80]: Audio: ac3, 48000 Hz, mono, fltp, 192 kb/s
This shows there are 4 audio streams. When I play this file in VLC / QuickTime it seems that Audio Track 4 is the default. I'd like to understand how this is chosen. Is it something within the mpg container format or are players choosing the stream that has the lowest id (0x80) ?
More background, when I try to turn this into a mp4 file with the following command:
ffmpeg -i da-orig.mpg -c copy -map 0 da-copy.mp4
I get roughly the same size file, but the default audio track is stream #0:1[0x85].
What I want is an equivalent mp4 file (so the same audio track chosen).
as written in this guide
"The -map option is used to choose which streams from the input(s)
should be included in the output(s)."
in your case you have only one input so it would be -map 0
if you have 2 inputs and want video from one and audio from the other it would be -map 0:v -map 1:a
since your input is a container selecting the video would be -map 0:v
and the second audio stream would be -map 0:a:2
ffmpeg -i da-orig.mpg -map 0:v -c:v h264 -crf 17 -preset 'veryslow' -map 0:a:2 -c:a copy output.mp4
to answer to your comment the sequence can help you
after you rename your collection with sequential numbers 'collection-name_0000' then
ffmpeg -i collection_name_#04d.mpg -map 0:v -c:v h264 -crf 17 -preset 'veryslow' -map 0:a:2 -c:a copy output-#04d.mp4
this iterate through the videos if they have the same number of streams
I'm having an issue with ffmpeg 3.2.2; ordinarily I ask it to make an MP4 video file with 2 audio streams. The command line looks like this:
ffmpeg.exe
-rtbufsize 256M
-f dshow -i video="screen-capture-recorder" -thread_queue_size 512
-f dshow -i audio="Line 2 (Virtual Audio Cable)"
-f dshow -i audio="Line 3 (Virtual Audio Cable)"
-map 0:v -map 1:a -map 2:a
-af silencedetect=n=-50dB:d=60 -pix_fmt yuv420p -y "c:\temp\2channelvideo.mp4"
I've wrapped it for legibility. This once worked fine, but something is wrong lately - it doesnt seem to record any audio, even though I can use other tools like Audacity to record audio from these devices just fine
I'm trying to do some diag on it by dropping the video component and asking ffmpeg to record the two audio devices to two separate files:
ffmpeg.exe
-f dshow -i audio="Line 2 (Virtual Audio Cable)" "c:\temp\line2.mp3"
-f dshow -i audio="Line 3 (Virtual Audio Cable)" "c:\temp\line3.mp3"
ffmpeg's console output looks like:
Guessed Channel Layout for Input Stream #0.0 : stereo
Input #0, dshow, from 'audio=Line 2 (Virtual Audio Cable)':
Duration: N/A, start: 5935.810000, bitrate: 1411 kb/s
Stream #0:0: Audio: pcm_s16le, 44100 Hz, stereo, s16, 1411 kb/s
Guessed Channel Layout for Input Stream #1.0 : stereo
Input #1, dshow, from 'audio=Line 3 (Virtual Audio Cable)':
Duration: N/A, start: 5936.329000, bitrate: 1411 kb/s
Stream #1:0: Audio: pcm_s16le, 44100 Hz, stereo, s16, 1411 kb/s
Output #0, mp3, to 'c:\temp\line2.mp3':
Metadata:
TSSE : Lavf57.56.100
Stream #0:0: Audio: mp3 (libmp3lame), 44100 Hz, stereo, s16p
Metadata:
encoder : Lavc57.64.101 libmp3lame
Output #1, mp3, to 'c:\temp\line3.mp3':
Metadata:
TSSE : Lavf57.56.100
Stream #1:0: Audio: mp3 (libmp3lame), 44100 Hz, stereo, s16p
Metadata:
encoder : Lavc57.64.101 libmp3lame
Stream mapping:
Stream #0:0 -> #0:0 (pcm_s16le (native) -> mp3 (libmp3lame))
Stream #0:0 -> #1:0 (pcm_s16le (native) -> mp3 (libmp3lame))
Press [q] to stop, [?] for help
The problem i'm currently having is that the produced mp3 are identical copies of line 2 only; line 3 audio is not recorded. The last line is of concern; it seems to be saying that stream 0 is being mapped to both output 0 and 1? Do I need a map command for each file also? I thought it would be implicit due to the way i specified the arguments
Turned out I needed to add a -map x:a between each source and output file, where x was either 0 or 1 depending on if it was the first or second source..
Want to batch convert a bunch of different video files from cli instead of Rolands old-and-slow-drag-and-drop-one-file-at-a-time-software. I have used ffprobe in OS X Terminal here. This shows us what the software did to the file and I want to do the same. MJPEG AVI I get but the rest, how would my ffmpeg syntax look to achieve this result efter converting?
Example: My ffprobe give me this
Input #0, avi, from 'P10_0001.AVI':
Metadata:
comment :
encoder : Roland Corporation
Duration: 00:03:17.64, start: 0.000000, bitrate: 16694 kb/s
Stream #0:0: Video: mjpeg (MJPG / 0x47504A4D), yuvj422p(pc, bt470bg/unknown/unknown), 640x480, 15285 kb/s, 25 fps, 25 tbr, 25 tbn, 25 tbc
Stream #0:1: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, 2 channels, s16, 1411 kb/s
What would the ffmpeg syntax look like to do this with a new file.
I've been trying some simple ones but those are not accepted by the machine (Edirol p-10) and I hope someone can point me in the right direction. :)
Edit:
OK. The syntax I want to do is involving 3 files.
File that has the correct codec and everything to work with the machine. P10_0001.AVI
A file that does not have the correct format (codec etc.) softvision.mpg
A new file just as file 2 but with the codec of file number 1. P10_0002.AVI
ffmpeg -i gradomat.mpg -framerate 25 -vf scale=640:480 -vcodec mjpeg -pix_fmt yuvj422p -b:v 15285k -b:a 1411k -acodec pcm_s16le -ar 44100 -ac 2 -metadata encoder="Roland Corporation" P10_000X.AVI
Think this solved it temporarily but the problem is that I have to write that my self, it would have been better if ffprobe gave me that syntax instead.
This is also a solution, but in python.
https://github.com/cskonopka/rolandp10fp
Is there any way to use ffmpeg to accurately break audio files into smaller files of a specific file size, or pull a specific number of samples from a file?
I'm working with a speech-to-text API that needs audio chunks in exactly 160,000 bytes, or 80,000 16-bit samples.
I have a video stream, and I have an ffmpeg command to extract audio from it:
ffmpeg -i "rtmp://MyFMSWorkspace/ingest/test/mp4:test_1000 live=1" -ar 16000 -f segment -segment_time 10 out%04d.wav
So now I have ~10 second audio chunks with a sample rate of 16 kHz. Is there any way to break this into exactly 160kb, 5 second files using ffmpeg?
I tried this:
ffmpeg -t 00:00:05.00 -i out0000.wav outCropped.wav
But the output was this:
Input #0, wav, from 'out0000.wav':
Metadata:
encoder : Lavf56.40.101
Duration: 00:00:10.00, bitrate: 256 kb/s
Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 16000 Hz, 1 channels, s16, 256 kb/s
Output #0, wav, to 'outCropped.wav':
Metadata:
ISFT : Lavf56.40.101
Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 16000 Hz, mono, s16, 256 kb/s
Metadata:
encoder : Lavc56.60.100 pcm_s16le
Stream mapping:
Stream #0:0 -> #0:0 (pcm_s16le (native) -> pcm_s16le (native))
Press [q] to stop, [?] for help
size= 156kB time=00:00:05.00 bitrate= 256.1kbits/s
but now the size is 156kb
EDIT:
My finished command is:
ffmpeg -i "url" -map 0:1 -af aresample=16000,asetnsamples=16000 -f segment -segment_time 5 -segment_format sw out%04d.sw
That output looks perfectly right. That ffmpeg size is expressed in KiB although it says kB. 160000 bytes = 156.25 kB + some header data. ffmpeg shows size with fractional part hidden. If you want a raw file, with no headers, output to .raw instead of .wav.
For people converting video files to MP3s split into 30 minute segments:
ffmpeg -i "something.MP4" -q:a 0 -map a -f segment -segment_time 1800 FileNumber%04d.mp3
The -q option can only be used with libmp3lame and corresponds to the LAME -V option (source)