[FFmpeg-user] wav concat woes

[Sean Darcy]

On Fedora 20, trying to concatenate a set of wav files.

cat 01_list.txt
file '01/01.01_test.wav'
file '01/01.02_test.wav'
file '01/01.03_test.wav'
file '01/01.04_test.wav'

ffmpeg  -f concat -i 01_list.txt -vn -c:a libfdk_aac    -profile:a 
aac_he -b:a 48k -ac 1 -signaling implicit    test2.m4a 2> ffmpeg.out

ffmpeg version 2.2.git Copyright (c) 2000-2014 the FFmpeg developers
   built on Apr  2 2014 11:12:31 with gcc 4.8.2 (GCC) 20131212 (Red Hat 
4.8.2-7)
   configuration: --prefix=/usr --bindir=/usr/bin 
--datadir=/usr/share/ffmpeg --libdir=/usr/lib64 --mandir=/usr/share/man 
--shlibdir=/usr/lib64 --extra-cflags='-Ofast -march=native -mtune=native 
-fopenmp -fomit-frame-pointer -pipe' --enable-avresample --enable-static 
--enable-shared --enable-gray --enable-gpl --enable-nonfree 
--enable-version3 --enable-postproc --enable-avfilter 
--enable-avresample --enable-pthreads --enable-x11grab --enable-gray 
--enable-vaapi --enable-hardcoded-tables --enable-libaacplus 
--enable-avisynth --enable-frei0r --enable-libfdk-aac --enable-libgsm 
--enable-libmp3lame --enable-libopenjpeg --enable-libopus 
--enable-librtmp --enable-libschroedinger --enable-libspeex 
--enable-libtheora --enable-libvo-aacenc --enable-libvorbis 
--enable-libvpx --enable-libwebp --enable-libx264 --enable-libxvid 
--enable-zlib --disable-debug --cpu=amdfam10 --arch=x86_64 --enable-pic 
--enable-libopencv --enable-openssl
   libavutil      52. 71.100 / 52. 71.100
   libavcodec     55. 56.107 / 55. 56.107
   libavformat    55. 36.100 / 55. 36.100
   libavdevice    55. 11.100 / 55. 11.100
   libavfilter     4.  3.100 /  4.  3.100
   libavresample   1.  2.  0 /  1.  2.  0
   libswscale      2.  5.102 /  2.  5.102
   libswresample   0. 18.100 /  0. 18.100
   libpostproc    52.  3.100 / 52.  3.100
[concat @ 0x82de60] Estimating duration from bitrate, this may be inaccurate
Guessed Channel Layout for  Input Stream #0.0 : stereo
Input #0, concat, from '01_list.txt':
   Duration: 00:00:00.00, start: 0.000000, bitrate: 1410 kb/s
     Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, 
stereo, s16, 1411 kb/s
Output #0, ipod, to 'test2.m4a':
   Metadata:
     encoder         : Lavf55.36.100
     Stream #0:0: Audio: aac (libfdk_aac) (HE-AAC) (mp4a / 0x6134706D), 
44100 Hz, mono, s16, 48 kb/s
Stream mapping:
   Stream #0:0 -> #0:0 (pcm_s16le -> libfdk_aac)
Press [q] to stop, [?] for help
size=     281kB time=00:00:47.81 bitrate=  48.2kbits/s    ^Msize= 
559kB time=00:01:35.27 bitrate=  48.1kbits/s    ^Msize=     838kB 
time=00:02:22.82 bitrate=  48.1kbits/s    ^Msize=    1115kB 
time=00:03:10.05 bitrate=  48.0kbits/s    ^Msize=    1390kB 
time=00:03:57.14 bitrate=  48.0kbits/s    ^Msize=    1667kB 
time=00:04:44.28 bitrate=  48.0kbits/s    ^MDTS -406750698222075776, 
next:301253333 st:0 invalid dropping
PTS -406750698222075776, next:301253333 invalid dropping st:0
DTS -406750698222074752, next:301276552 st:0 invalid dropping
PTS -406750698222074752, next:301276552 invalid dropping st:0
DTS -406750698222073728, next:301299771 st:0 invalid dropping
.............


The other wav files have the same parameters:

ffprobe 01/01.02_test.wav
ffprobe version 2.2.git Copyright (c) 2007-2014 the FFmpeg developers
......................
Input #0, wav, from '01/01.02_test.wav':
   Duration: 00:05:00.13, bitrate: 1411 kb/s
     Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, 2 
channels, s16, 1411 kb/s

And I can run the command on each individually.

Any help appreciated.

sean