[FFmpeg-user] Concatenating two audio files

Anatol anatol2002 at gmail.com
Thu Jul 31 13:40:36 CEST 2014


Turn your WAV files into raw PCM files -
ffmpeg -i INPUT1 -c:a copy -vn -f s16le -y PCM1

Then u can either cat them together to get a raw PCM file or use ffmpeg
concat protocol:
ffmpeg -ar SAMPLE_RATE -channel_layout STEREO/MONO -f s16le -i
concat:"PCM1|PCM2|..." -vn -c:a copy -f wav -y OUTPUT


On Thu, Jul 31, 2014 at 1:47 PM, Stuart Bruce <stuartrbruce at gmail.com>
wrote:

> Hello
>
> I've been trying to get FFmpeg to concatenate two audio files together,
> based on the top answer from this forum:
>
> http://superuser.com/questions/587511/concatenate-multiple-wav-files-using-single-command-without-extra-file
>
> Although I believe I've replicated the winning answer, I can't seem to work
> out what the correct syntax is to make it work.
>
> My log output (original parameters and console output) is below. For
> security I've had to change all the actual file paths to '********' but all
> the paths and files are valid.
>
> I'm hoping somebody could point me to the problem, which I'm hoping is
> something as simple as having a quote mark out of place or something like
> that. Any help would be very much appreciated.
>
> Thanks
>
> Stuart
>
>
> FFmpeg process log.
> Started: 31/07/2014 11:45:59
> exe:
> C:\Users\Stuart\*******\FFmpeg\ffmpeg.exe
> Arguments:
> -i
> "C:\Users\Stuart\AppData\*********\cache\310714_3Vc1AQqWBXtqimH5Sj1j.wav"
> -i
> "C:\Users\Stuart\AppData\**********\cache\310714_bGhog6vOWMmn8x4J8ADz.wav"
> -filter_complex '[0:0][1:0]concat=n=2:v=0:a=1[out]' -map [out]
> "C:\Users\Stuart\******\s-tk-220-00200.wav"
> FFmpeg output:
> ffmpeg version N-53868-gc51654f Copyright (c) 2000-2013 the FFmpeg
> developers
>   built on Jun  6 2013 03:00:45 with gcc 4.7.3 (GCC)
>   configuration: --enable-gpl --enable-version3 --disable-w32threads
> --enable-avisynth --enable-bzlib --enable-fontconfig --enable-frei0r
> --enable-gnutls --enable-iconv --enable-libass --enable-libbluray
> --enable-libcaca --enable-libfreetype --enable-libgsm --enable-libilbc
> --enable-libmodplug --enable-libmp3lame --enable-libopencore-amrnb
> --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus
> --enable-librtmp --enable-libschroedinger --enable-libsoxr
> --enable-libspeex --enable-libtheora --enable-libtwolame
> --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis
> --enable-libvpx --enable-libx264 --enable-libxavs --enable-libxvid
> --enable-zlib
>   libavutil      52. 35.100 / 52. 35.100
>   libavcodec     55. 15.100 / 55. 15.100
>   libavformat    55.  8.102 / 55.  8.102
>   libavdevice    55.  2.100 / 55.  2.100
>   libavfilter     3. 75.101 /  3. 75.101
>   libswscale      2.  3.100 /  2.  3.100
>   libswresample   0. 17.102 /  0. 17.102
>   libpostproc    52.  3.100 / 52.  3.100
> Guessed Channel Layout for  Input Stream #0.0 : mono
> Input #0, wav, from
> 'C:\Users\Stuart\*****************\cache\310714_3Vc1AQqWBXtqimH5Sj1j.wav':
>   Metadata:
>     encoder         : Lavf55.8.102
>   Duration: 00:00:04.18, bitrate: 705 kb/s
>     Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, mono,
> s16, 705 kb/s
> [wav @ 00353400] max_analyze_duration 5000000 reached at 5015510
> microseconds
> [wav @ 00353400] Estimating duration from bitrate, this may be inaccurate
> Guessed Channel Layout for  Input Stream #1.0 : stereo
> Input #1, wav, from
> 'C:\Users\Stuart\**********\cache\310714_bGhog6vOWMmn8x4J8ADz.wav':
>   Duration: 00:00:05.20, bitrate: 1411 kb/s
>     Stream #1:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz,
> stereo, s16, 1411 kb/s
> [AVFilterGraph @ 0035bd80] No such filter:
> '[0:0][1:0]concat=n=2:v=0:a=1[out]'
> Error configuring filters.
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