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
Related
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
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
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've been trying for a few hours to get the right arguments to do the below operation with ffmpeg, but i'm a bit out of my depth. If anyone can give me some guidance, that would be very helpful.
Input file is MXF with one video track and two separate 48k 24b PCM audio tracks.
I want to extract just the audio, map first track to left, second track to right and output as a stereo wave file.
I am trying variations on the below command, but it seems to be unhappy about the video stream, giving the error
"mapchan: stream #0.0 is not an audio stream."
ffmpeg -i test2.mxf -vn -map_channel 0.0.1 output.wav_CH0 -map_channel 0.0.2 output.wav_CH1
Here is ffmpeg's analysis of the streams in case it is helpful:
Stream #1:0: Video: mpeg2video (4:2:2), yuv422p(tv, bt709), 1920x1080 [SAR 1:1 DAR 16:9], max. 50000 kb/s, 25 fps, 25 tbr, 25 tbn, 50 tbc
Stream #1:1: Audio: pcm_s24le, 48000 Hz, mono, s32, 1152 kb/s
Stream #1:2: Audio: pcm_s24le, 48000 Hz, mono, s32, 1152 kb/s
Stream #1:3: Audio: pcm_s24le, 48000 Hz, mono, s32, 1152 kb/s
Stream #1:4: Audio: pcm_s24le, 48000 Hz, mono, s32, 1152 kb/s
Stream #1:5: Audio: pcm_s24le, 48000 Hz, mono, s32, 1152 kb/s
Stream #1:6: Audio: pcm_s24le, 48000 Hz, mono, s32, 1152 kb/s
Stream #1:7: Audio: pcm_s24le, 48000 Hz, mono, s32, 1152 kb/s
Stream #1:8: Audio: pcm_s24le, 48000 Hz, mono, s32, 1152 kb/s
The filtergraph function seems to do the trick.
ffmpeg -i test2.mxf -filter_complex "[0:1:0][0:2:0]amerge=inputs=2[aout]" -map "[aout]" output.wav
The 0:1:0 seems to represent input1 (the -i option), stream1 (the track), channel (left/right if it were stereo). --could be wrong with my interpretation here, but it works nonetheless.
ffmpeg documentation and FFmpeg Wiki: Audio Channel Manipulation were very helpful.