In raspberry pi I've following i2s microphone breakout board and use it like the guide suggested. When I try record audio from it using ffmpeg to the file with ffmpeg -f alsa -i dmic_sv out.wav command. I'll receive following error
[alsa # 0x22e21c0] cannot set sample format 0x10000 2 (Invalid argument)
dmic_sv: Input/output error
When I specify the used codec explicitly with -acodec it works fine:
ffmpeg -f alsa -acodec pcm_s32le -i dmic_sv out.wav
And from the output ffmpeg will reencode to pcm_s16le
Input #0, alsa, from 'dmic_sv':
Duration: N/A, start: 1597597938.887969, bitrate: 3072 kb/s
Stream #0:0: Audio: pcm_s32le, 48000 Hz, stereo, s32, 3072 kb/s
Stream mapping:
Stream #0:0 -> #0:0 (pcm_s32le (native) -> pcm_s16le (native))
How I can tell ffmpeg to use signed 32-bit little endian by default without specifying it explicitly? And where ffmpeg gets this default and can I configure it somehow?
I figured this one out myself by reading ffmpeg source code. It seems when codec is not specified and alsa device is used. FFmpeg will default to pcm 16-bit samples instead. Code to set the default here and the default macro here.
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
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
If I don't make a mistake, Safari currently need MP4 (H.264/AAC) video encoded for the HTML5 <video> element.
So I tried to convert a video to this format with ffmpeg. However when I enter the shell command ffmpeg -i video.flv video.mp4, the returned error is :
Seems stream 0 codec frame rate
differs from container frame rate:
2000.00 (2000/1) -> 29.92 (359/12) Input #0, flv, from 'video.flv':
Duration: 00:05:01.20, start:
0.000000, bitrate: 66 kb/s
Stream #0.0: Video: h264, yuv420p, 320x240 [PAR 1:1 DAR 4:3], 66 kb/s,
29.92 tbr, 1k tbn, 2k tbc
Stream #0.1: Audio: aac, 22050 Hz, stereo, s16 Output #0, mp4, to
'video.mp4':
Stream #0.0: Video: mpeg4, yuv420p, 320x240 [PAR 1:1 DAR 4:3],
q=2-31, 200 kb/s, 90k tbn, 29.92 tbc
Stream #0.1: Audio: 0x0000, 22050 Hz, stereo, s16, 64 kb/s Stream
mapping: Stream #0.0 -> #0.0
Stream #0.1 -> #0.1 Unsupported codec
for output stream #0.1
An AAC codec is required but I'm quite newbie with ubuntu and I dont really now how to fix this problem. I'm using Ubuntu 9.10 Karmik Koala (for amd64).
Thank you very much. :)
http://handbrake.fr is a nice high level tool with a lot of useful presets for mp4 for iPod, PS3, ... with both GUI and CLI interfaces for Linux, Windows and Mac OS X.
It comes with its own dependencies as a single statically linked fat binary so you have all the x264 / aac codecs included.
$ HandBrakeCLI -Z Universal -i myinputfile.mov -o myoutputfile.mp4
To list all the available presets:
$ HandBrakeCLI -z
Software patents led Debian/Ubuntu to disable the H.264 and AAC encoders in ffmpeg. See /usr/share/doc/ffmpeg/README.Debian.gz.
So go install x264, mplayer/mencoder, and Nero's AAC encoder. (Or, if you want to use all Free software, and don't care so much about audio quality, then sudo aptitude install faac.)
I don't remember if the medibuntu package of mencoder includes x264 vid encoding, since I build my own from git x264 and svn mplayer sources. (x264 is very actively developed, with significant quality and speed improvements frequently added.)
http://git.videolan.org/?p=x264.git;a=summary
x264 is also packaged, but you should check that it's up to date enough to include weightp with recent bugfixes, and even more recent speed improvements...
Or if you're already willing to convert from .flv, instead of going from the high-quality source the flv was made from, then probably whatever recent version of x264 you can find will be fine.
You're trying to convert a (rather rare) .flv file that (already) contains H.264 video and AAC audio.
Formatting your console's output as FFmpeg brings out these details.
Input #0, flv, from 'video.flv':
Duration: 00:05:01.20, start: 0.000000, bitrate: 66 kb/s
Stream #0.0: Video: h264, yuv420p, 320x240 [PAR 1:1 DAR 4:3], 66 kb/s, 29.92 tbr, 1k tbn, 2k tbc
Stream #0.1: Audio: aac, 22050 Hz, stereo, s16
The original flv is converted to an .mp4 file with H.264 video and AAC audio (just like the original .flv):
Output #0, mp4, to 'video.mp4':
Stream #0.0: Video: mpeg4, yuv420p, 320x240 [PAR 1:1 DAR 4:3], q=2-31, 200 kb/s, 90k tbn, 29.92 tbc
Stream #0.1: Audio: 0x0000, 22050 Hz, stereo, s16, 64 kb/s
Because the audio and video data in the .flv are already in the format/codecs you need for the .mp4, you can just copy everything to the new .mp4 container. This process will be massively faster than decoding and reencoding everything:
ffmpeg -i video.flv -vcodec copy -acodec copy video.mp4
or more simply:
ffmpeg -i video.flv -codec copy video.mp4
##The real error you're getting is:##
Unsupported codec for output stream #0.1
Which means FFmpeg can't convert audio (stream #0.1) to AAC.
You can skip the error by:
copying the audio data since it's already AAC encoded (use the copy command above)
or you can solve the error by:
using a FFmpeg build with AAC decode/encode support. FFmpeg currently supports 4 AAC libraries (see FFmpeg and AAC Encoding Guide).
For more details you should also read Converting FLV to MP4 With FFmpeg The Ultimate Guide
You need to recompile ffmpeg (from source) so that it supports x264. If you follow the instructions in this page, then you will be able to peform any kind of conversion you want.
You can also try adding the Motumedia PPA to your apt sources and update your ffmpeg packages.
Had this problem recently with converting nasty WMV into Final Cut Pro X for editing. Flow player can do it but it leaves a water mark, so I fiddled a bit with ffmpeg till I got something going.
First install ffmpeg - I used
brew install ffmpeg
Obviously you need brew installed first, google that bit.
Next I wrote a simple command line script with the following content - you can substitute the $1 for an input / output file or just create a shell script file...
vi convert.sh
Paste.
echo "Pass one"
ffmpeg -y -i "$1" -c:v libx264 -preset medium -b:v 1555k -pass 1 -c:a libfaac -b:a 256k -f mp4 /dev/null &&
echo "Pass two"
ffmpeg -i "$1" -c:v libx264 -preset medium -b:v 1555k -pass 2 -c:a libfaac -b:a 256k "$1.mp4"
Then to convert your video...
sh convert.sh myvideofile.wmv
If all went well you should see a new file called myvideofile.wmv.mp4.
Hope that works for you.
You need to compile ffmpeg with an AAC encoder. You can find one at AudioCoding.
Try This one:: Libav in Linux
Installation: run command
sudo apt-get install libav-tools
Video conversion command::Go to folder contains the video and run in terminal
avconv -i oldvideo.flv -ar 22050 convertedvideo.mp4
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.