I am trying to concat two video files with ffmpeg concat demuxer for most of the part it works just fine!
But when I try to concat videos which has two different audio profile with same codec, it concats with the resulting video having weird sound problem. And when re-encoding the resulting video it will spit out a lots of error related to audio.
Here is ffprobe output for some audio stream from different video files
Video 1
[STREAM]
index=1
codec_name=aac
codec_long_name=unknown
profile=4
codec_type=audio
codec_time_base=1/48000
codec_tag_string=[0][0][0][0]
codec_tag=0x0000
sample_fmt=fltp
...
[/STREAM]
Video 2
[STREAM]
index=1
codec_name=aac
codec_long_name=unknown
profile=1
codec_type=audio
codec_time_base=1/48000
codec_tag_string=[0][0][0][0]
codec_tag=0x0000
sample_fmt=fltp
...
[/STREAM]
Video 3
[STREAM]
index=1
codec_name=aac
codec_long_name=unknown
profile=28
codec_type=audio
codec_time_base=1/48000
codec_tag_string=[0][0][0][0]
codec_tag=0x0000
sample_fmt=fltp
...
[/STREAM]
Look the different profile= values.I was able to reproduce 28 and 1 but was failed for 4
28 = he_aac_v2 1 = ffmpeg default
So what I want to know the most is,
What does these different values mean for aac?
And how to reproduce them with any aac encode?
According to libavcodec/avcodec.h:
FF_PROFILE_AAC_MAIN 0
FF_PROFILE_AAC_LOW 1
FF_PROFILE_AAC_SSR 2
FF_PROFILE_AAC_LTP 3
FF_PROFILE_AAC_HE 4
FF_PROFILE_AAC_HE_V2 28
FF_PROFILE_AAC_LD 22
FF_PROFILE_AAC_ELD 38
The native FFmpeg AAC encoder (-c:a aac) does not have the ability to output HE or HEv2 profiles.
If you need HE profile (-profile:a 4 or -profile:a aac_he) you'll have to use another encoder, such as -c:a libfdk_aac, -c:a aac_at (macOS/iOS only), or a separate standalone AAC encoder.
Related
I need to convert from .mp3 to .gsm (preferably with ffmpeg).
I used it for several different formats but with this isn't as simple as it was with the others.
I don't know what parameters I'm missing.
I tried using ffmpeg with the following comand:
ffmpeg -i ".\example.mp3" ".\example.gsm"
But it shows me the following error:
Sample rate 8000Hz required for GSM, got 44100Hz 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 Conversion failed!
ffmpeg -ar 8000 -ac 1 -i ".\example.mp3" ".\example.gsm"
-ar sample rate
-ac audio channel
I want to convert a MOV from my Casio cam to mp4 using transcode. Why transcode? Because I also want to deshake the video in the same step.
When I use
ffmpeg -i in.MOV out.mp4
it works. When using
transcode -J stabilize -i in.MOV
or
transcode -J transform -i in.MOV -y ffmpeg -F mpeg4 -o out.mp4
I get hundreds of these errors:
[ffmpeg_audio] Error: avcodec_open2 failed
[adpcm_ima_wav # 0x1f7f180] Only 4-bit ADPCM IMA WAV files are supported
This looks to me as if transcode uses ffmpeg internally.
I could use ffmpeg to make it mp4 first and then use transcode to stabilize the video, but then it would be re-encoded twice which I would like to avoid.
This is what mplayer says about my MOV file:
MPlayer2 2.0-701-gd4c5b7f-2ubuntu2 (C) 2000-2012 MPlayer Team
Cannot open file '/home/koem/.mplayer/input.conf': No such file or directory
Failed to open /home/koem/.mplayer/input.conf.
Cannot open file '/etc/mplayer/input.conf': No such file or directory
Failed to open /etc/mplayer/input.conf.
Playing 1-original.MOV.
Detected file format: QuickTime / MOV (libavformat)
[lavf] stream 0: video (h264), -vid 0
[lavf] stream 1: audio (adpcm_ima_wav), -aid 0, -alang eng
Clip info:
major_brand: qt
minor_version: 537921536
compatible_brands: qt caqv
creation_time: 2017-01-02 23:31:38
Load subtitles in .
Failed to open VDPAU backend libvdpau_i965.so: cannot open shared object file: No such file or directory
[vdpau] Error when calling vdp_device_create_x11: 1
[ass] auto-open
Selected video codec: H.264 / AVC / MPEG-4 AVC / MPEG-4 part 10 [libavcodec]
Selected audio codec: ADPCM IMA WAV [libavcodec]
AUDIO: 44100 Hz, 2 ch, s16le, 352.8 kbit/25.00% (ratio: 44100->176400)
AO: [pulse] 44100Hz 2ch s16le (2 bytes per sample)
Starting playback...
VIDEO: 1920x1080 29.970 fps 15940.0 kbps (1992.5 kB/s)
VO: [xv] 1920x1080 => 1920x1080 Planar YV12
Colorspace details not fully supported by selected vo.
A: 1.1 V: 1.1 A-V: -0.000 ct: 0.000 0/ 0 16% 8% 1.6% 0 0
Exiting... (Quit)
How can I make it work with transcode without using ffmpeg first?
FFmpeg has a deshake as well as a stabilization filter. Get a new binary if yours doesn't.
To continue with your existing binaries, run
ffmpeg -i in.MOV -vcodec copy out.mp4
This will skip video re-encoding.
I'm looking for a way to re-encode the video stream of a movie only and keep all other streams as they are using ffmpeg or more specific streamio/streamio-ffmpeg (Github - StreamIO-FFMPEG).
I already tried various combinations of -map 0 or -map a:0 -map s:0, but in all combinations I tried, either nothing is encoded at all, or not all other streams are copied to the new file. In most cases there is only one audio stream after encoding, when there were two before, and sometimes the subtitle streams are lost, too. Also most times the info what language the streams are in gets lost.
So when I have a movie file (mkv) with the following streams:
0: video [H.264, 1080p]
1: audio [english, mp3]
2: audio [french, mp3]
3: subtitle [english (forced)]
4: subtitle [english]
What should the ffmpeg parameters be, if I want to encode the video file to H.265 and 720p and keep all other streams as they are?
What should the parameters be, if I additionally want to encode the audio streams as AAC?
Thanks in advance!
Use
ffmpeg -i in.mkv -vf scale=hd720 -map 0 -c copy -c:v libx265 out.mkv
To encode audio as well,
ffmpeg -i in.mkv -vf scale=hd720 -map 0 -c copy -c:v libx265 -c:a aac out.mkv
(The order of the arguments above matter)
I have converted my .mkv file into an .mp4 by using the command:
sudo avconv -i input.mkv -codec copy output.mp4
I am trying to play the mp4 file in the browser, but the audio is not playing. The video player shows that it is on mute, but the button is disabled so you cannot turn it off of mute.
Other mp4s are working, but they were not converted from .mkv. Any help would be much appreciated.
In an MP4 container, browsers usually support only H.264 video and either AAC or MP3 audio. The output from your avconv command should show the format of your video and audio; look under "Input #0" for the lines that start with "Stream #". If you audio is not already AAC or MP3 you will want to convert it instead of just copying it to the MP4 container. You can copy the video and convert only the audio with a command like this:
avconv -i input.mkv -c:v copy -c:a libmp3lame -q:a 2 output.mp4
The -c:v copy will copy the video, and -c:a libmp3lame will convert the audio to MP3 using the libmp3lame encoder. -q:a 2 sets the audio quality; use a lower number for better quality (and a larger file). You could instead convert to AAC audio if your avconv was configured with non-free codecs enabled and a good quality AAC encoder.
I have an avi file that has different levels of audio. Is there a way to decrease and increase appropriately where needed the audio of my file using ffmpeg?
In ffmpeg you can use the volume filter to change the volume of a track. Make sure you download a recent version of the program.
Find out the gain to apply
First you need to analyze the audio stream for the maximum volume to see if normalizing would even pay off:
ffmpeg -i video.avi -af "volumedetect" -f null /dev/null
Replace /dev/null with NUL on Windows. This will output something like the following:
[Parsed_volumedetect_0 # 0x7f8ba1c121a0] mean_volume: -16.0 dB
[Parsed_volumedetect_0 # 0x7f8ba1c121a0] max_volume: -5.0 dB
[Parsed_volumedetect_0 # 0x7f8ba1c121a0] histogram_0db: 87861
As you can see, our maximum volume is -5.0 dB, so we can apply 5 dB gain. If you get a value of 0 dB, then you don't need to normalize the audio.
Apply the volume filter:
Now we apply the volume filter to an audio file. Note that applying the filter means we will have to re-encode the audio stream. What codec you want for audio depends on the original format, of course. Here are some examples:
Plain audio file: Just encode the file with whatever encoder you need:
ffmpeg -i input.wav -af "volume=5dB" output.mp3
Your options are very broad, of course.
AVI format: Usually there's MP3 audio with video that comes in an AVI container:
ffmpeg -i video.avi -af "volume=5dB" -c:v copy -c:a libmp3lame -q:a 2 output.avi
Here we chose quality level 2. Values range from 0–9 and lower means better. Check the MP3 VBR guide for more info on setting the quality. You can also set a fixed bitrate with -b:a 192k, for example.
MP4 format: With an MP4 container, you will typically find AAC audio. We can use ffmpeg's build-in AAC encoder.
ffmpeg -i video.mp4 -af "volume=5dB" -c:v copy -c:a aac -strict experimental -b:a 192k output.mp4
Here you can also use other AAC encoders. Some of them support VBR, too. See this answer and the AAC encoding guide for some tips.
In the above examples, the video stream will be copied over using -c:v copy. If there are subtitles in your input file, or multiple video streams, use the option -map 0 before the output filename.
The author's info is: Jon Skarpeteig in SuperUser
You can use my ffmpeg-normalize script for that.
First, install a recent version of ffmpeg. Then, install via pip install ffmpeg_normalize, then run it on an AVI file:
ffmpeg-normalize input.avi -o output.mkv -c:a aac -b:a 192k
Here, we're choosing to re-encode the audio with AAC at 192 kBit/s, and copy the video stream over to the output. This will perform EBU R128 normalization, but simple peak/RMS normalization is also possible.