If I run this command line
ffmpeg -ss 0 -t 3600 -i file1.mp3 -ss 0 -t 20 -i file2.mp3 -filter_complex "[0][1]concat=n=2:v=0:a=1" -ac 2 -f wav - > test.wav
I'm basically putting the stout inside a container wav (test.wav) but the duration is always wrong. The output file should be 01:00:20.00 but if I play it on VLC (or any player audio) it shows 06:12:49.00 and even if I change the start_times, the durations and number of files, I still get that timecode out. The even weirder thing is that ffprobe shows the duration as it should be. Can somebody please help me on this?
UPDATE:
[wav # 0000000000cf3680] Ignoring maximum wav data size, file may be invalid
[wav # 0000000000cf3680] Estimating duration from bitrate, this may be inaccurate
Input #0, wav, from 'test.wav':
Metadata:
encoder : Lavf57.72.101
timecode : 01:00:20.00
Duration: 01:00:20.00, bitrate: 1536 kb/s
Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 48000 Hz, 2 channels, s16, 1536 kb/s
That is what the ffprobe on the output shows..the duration is correct here but not on any audio player
Related
I have two MP3 files that were created from the same source, with different audio within them. Here are the properties from ffprobe
Duration: 00:00:08.86, bitrate: 384 kb/s
Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 24000 Hz, 1 channels, s16, 384 kb/s
NOTE: Even though the file is an MP3 it shows as pcm_s16le
When I try and join the two files together using
ffmpeg -i download.mp3 -i download1.mp3 -filter_complex [0:a:0][1:a:0]concat=n=2:v=0:a=1[outa] -map [outa] joineddownloads.mp3
I get the following result and a big drop in bitrate(quality)
Duration: 00:00:10.42, start: 0.046042, bitrate: 32 kb/s
Stream #0:0: Audio: mp3, 24000 Hz, mono, fltp, 32 kb/s
How can I maintain the high 320kbs bitrate and all the other properties that were present before I created the joined file?
To avoid re-encoding - concatenate the two mp3s
First create a text file ‘files.txt’ containing two lines:
file '/path/download.mp3'
file '/path/download1.mp3'
Second:
ffmpeg -f concat -i files.txt -c copy out.mp3
I'm trying to extract audio tracks from some Avi videos and save them to their own files, ideally without re-encoding.
I've had a look through here https://www.ffmpeg.org/ffmpeg.html#Audio-Options and here ffmpeg to extract audio from video though I'm getting errors regardless of the approach I try.
My latest command string is:
ffmpeg -i /home/d/Pictures/Test/input-video.AVI -map 0:a -vn -acodec copy /home/d/Pictures/Test/output-audio.m4a
The key part of the output is:
Guessed Channel Layout for Input Stream #0.1 : mono
Input #0, avi, from '/home/d/Pictures/Test/input-video.AVI':
Duration: 00:00:05.94, start: 0.000000, bitrate: 18131 kb/s
Stream #0:0: Video: mjpeg (MJPG / 0x47504A4D), yuvj422p(pc, bt470bg/unknown/unknown), 1280x720, 17995 kb/s, 30.28 fps, 30.28 tbr, 30.28 tbn, 30.28 tbc
Stream #0:1: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 11025 Hz, 1 channels, s16, 176 kb/s
File '/home/d/Pictures/Test/output-audio.m4a' already exists. Overwrite ? [y/N] y
[ipod # 0x1d89520] Codec for stream 0 does not use global headers but container format requires global headers
[ipod # 0x1d89520] Could not find tag for codec pcm_s16le in stream #0, codec not currently supported in container
Output #0, ipod, to '/home/d/Pictures/Test/output-audio.m4a':
Metadata:
encoder : Lavf56.40.101
Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 11025 Hz, mono, 176 kb/s
Stream mapping:
Stream #0:1 -> #0:0 (copy)
Could not write header for output file #0 (incorrect codec parameters ?): Invalid argument
I'm believe I have got the right audio stream number from this output and thus am assuming the "-map 0:a" part isn't the problem.
I'm running on Linux Mint 18.1
MP4 family of formats don't store PCM audio, so you either have to re-encode or save to another format, like Matroska.
ffmpeg -i video.AVI -map 0:a -vn -acodec copy audio.mka
I am trying to create a four-channel mp4 file with AAC encoding for ambisonics use. I am trying to encode a 4-channel first-order ambisonic wav file into AAC like so:
avconv -i four_channel_input.wav -c:a libfaac -ac 4 four_channel_output.mp4
This gives me the error
[libfaac # 0x7f938885a000] Specified channel_layout is not supported
Error initializing output stream 0:0 -- Error while opening encoder for output stream #0:0 - maybe incorrect parameters such as bit_rate, rate, width or height
Removing the -ac 4 option gives me a 5 channel file
Duration: 00:01:21.09, start: 0.021333, bitrate: 218 kb/s
Stream #0:0(und): Audio: aac (LC) [mp4a / 0x6134706D]
48000 Hz, 5.0, fltp, 215 kb/s (default)
with a blank first channel, which is obviously suboptimal. In order to create compressed ambisonics files, should I be using a separate format like AmbiX (even though I believe this is uncompressed)?
With ffmpeg, you can run
ffmpeg -i input.wav -c:a aac -ac 4 -channel_layout 4.0 four_channel_output.mp4
I have 1 audio file from dash stream
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'file_191282-377206_header.mp4': Metadata:
major_brand : iso6
minor_version : 1
compatible_brands: mp42dashmsdhmsixiso6avc1isom
creation_time : 2016-04-29T11:04:26.000000Z Duration: 00:00:30.02, start: 14.997333, bitrate: 49 kb/s
Stream #0:0(eng): Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz, stereo, fltp, 48 kb/s (default)
Also in exiftool output:
Movie Data Size : 180193
Movie Data Offset : 6388
I try to merge with another audio and save metadata info like Movie Data Size/Offset and start/duration time.
I try with ffmpeg/MP4Box commands like:
MP4Box -new -add file_191282-377206_header.mp4 -add out000.mp4
ffmpeg -i file_191282-377206_header.mp4 -i out000.mp4 -codec copy -shortest output.mp4
ffmpeg -i file_191282-377206_header.mp4 -i out000.mp4 -filter_complex amerge -ac 2 -c:a aac output.mp4
All the time information erased or changed.
So question is how to merge 2 audio files inside mp4 and not change Movie Size/Offset and start/duration time?
Try to merge the actual audio (AAC) not the media container (MP4).
So extract the AAC from each media file, then merge those audios.
Finalise by putting the merged audio back into a new MP4 output.
I have looked everywhere to find a linux utility that will allow me to download rtmp streams. Not flv video but MP3 streams. The location of the streams I want to download are in this format.
rtmp://live.site.com/loc/45/std_fc74a6b7f79c70a5f60.mp3
Anyone know of such a command line tool? Or even anything close to what I am asking for?
I do not want full software applications and it would be great if it worked on Linux via Shell or something.
Thanks all
One of the following should do, if you have mplayer or vlc compiled with RTMP access.
mplayer -dumpstream rtmp://live.site.com/loc/45/std_fc74a6b7f79c70a5f60.mp3
This will generate a ./stream.dump.
vlc -I dummy rtmp://live.site.com/loc/45/std_fc74a6b7f79c70a5f60.mp3 \
--sout file/ts:output.mpg vlc://quit
This will generate a ./output.mpg. You'll have to demux it to extract just the audio stream out.
This question is old but this can help to another users with this doubt.
To download directly, without any conversion, there is two options (the author of both programs is the same and the behavior is the same):
RTMPDump. Example: rtmpdump -r "rtmp://host.com/dir/file.flv" -o filename.flv
flvstreamer. Example: flvstreamer -r "rtmp://od.flash.plus.es/ondemand/14314/plus/plustv/PO770632.flv" -o salida.flv
And if you want download and convert the video at same time, the best way is use ffmpeg:
ffmpeg -i rtmp://server/live/streamName -acodec copy -vcodec copy dump.mp4
I think the landscape has changed a bit since the time of some of the previous answers. At least according to the rtmp wikipedia page. It would appear that the rtmp protocol specification is open for public use. To that end you can use 2 tools to accomplish what the original poster was asking, rtmpdump and ffmpeg. Here's what I did to download a rtmp stream that was sending an audio podcast.
step #1 - download the stream
I used the tool rtmpdump to accomplish this. Like so:
% rtmpdump -r rtmp://url/to/some/file.mp3 -o /path/to/file.flv
RTMPDump v2.3
(c) 2010 Andrej Stepanchuk, Howard Chu, The Flvstreamer Team; license: GPL
Connecting ...
INFO: Connected...
Starting download at: 0.000 kB
28358.553 kB / 3561.61 sec
Download complete
step #2 - convert the flv file to mp3
OK, so now you've got a local copy of the stream, file.flv. You can use ffmpeg to interrogate the file further and also to extract just the audio portion.
% ffmpeg -i file.flv
....
[flv # 0x25f6670]max_analyze_duration reached
[flv # 0x25f6670]Estimating duration from bitrate, this may be inaccurate
Input #0, flv, from 'file.flv':
Duration: 00:59:21.61, start: 0.000000, bitrate: 64 kb/s
Stream #0.0: Audio: mp3, 44100 Hz, 1 channels, s16, 64 kb/s
From the above output we can see that the file.flv contains a single stream, just audio, and it's in mp3 format, and it's a single channel. To extract it to a proper mp3 file you can use ffmpeg again:
% ffmpeg -i file.flv -vn -acodec copy file.mp3
....
[flv # 0x22a6670]max_analyze_duration reached
[flv # 0x22a6670]Estimating duration from bitrate, this may be inaccurate
Input #0, flv, from 'file.flv':
Duration: 00:59:21.61, start: 0.000000, bitrate: 64 kb/s
Stream #0.0: Audio: mp3, 44100 Hz, 1 channels, s16, 64 kb/s
Output #0, mp3, to 'file.mp3':
Metadata:
TSSE : Lavf52.64.2
Stream #0.0: Audio: libmp3lame, 44100 Hz, 1 channels, 64 kb/s
Stream mapping:
Stream #0.0 -> #0.0
Press [q] to stop encoding
size= 27826kB time=3561.66 bitrate= 64.0kbits/s
video:0kB audio:27826kB global headers:0kB muxing overhead 0.000116%
The above command will copy the audio stream into a file, file.mp3. You could also have extracted it to a wav file like so:
ffmpeg -i file.flv -vn -acodec pcm_s16le -ar 44100 -ac 2 file.wav
This page was useful in determining how to convert the flv file to other formats.