Volume adjust and channel merge on video using FFMPEG - audio

Question

I'm trying to convert a recorded AVI (1 video stream, 2 audio streams[1st is stereo, 2nd is mono]) video file to H264/AAC.
I want the second audio stream to be at 60% volume and the first at 100%.
I also want to merge the first and second audio stream.
Output should be H264 with AAC audio.
The command I tried to use is:
"ffmpeg.exe"
-i "input.avi"
-filter_complex "
[0:a:0]aformat=channel_layouts=stereo,volume=1.0[a1];
[0:a:1]aformat=channel_layouts=mono,volume=0.6[a2];
[a1][a2]amerge,pan=stereo|c0<c0+c2|c1<c1+c2[out]
"
-map 0:v
-map "[out]"
-c:v libx264 -preset slow -crf 26
-c:a aac -strict experimental -ab 128000 -ac 2 -ar 48000
"output.mp4"
But I get the error:
[AVFilterGraph # 0000000ceb630c00] The following filters could not choose their formats: Parsed_amerge_4
Consider inserting the (a)format filter near their input or output.
Error configuring complex filters.
Error number -5 occurred
What am I doing wrong and how can it be fixed?
Complete Console Log:
ffmpeg version N-76456-g6df2c94 Copyright (c) 2000-2015 the FFmpeg developers
built with gcc 5.2.0 (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-libbs2b --enable-libcaca --enable-libdcadec --enable-libfreetype --enable-libgme --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-libvidstab --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxavs --enable-libxvid --enable-lzma --enable-decklink --enable-zlib
libavutil 55. 5.100 / 55. 5.100
libavcodec 57. 14.100 / 57. 14.100
libavformat 57. 14.100 / 57. 14.100
libavdevice 57. 0.100 / 57. 0.100
libavfilter 6. 14.101 / 6. 14.101
libswscale 4. 0.100 / 4. 0.100
libswresample 2. 0.100 / 2. 0.100
libpostproc 54. 0.100 / 54. 0.100
[avi # 000000b1cfa6bc20] non-interleaved AVI
Guessed Channel Layout for Input Stream #0.1 : stereo
Guessed Channel Layout for Input Stream #0.2 : mono
Input #0, avi, from 'input.avi':
Metadata:
encoder : DxtoryCore ver2.0.0.122
ISRC : Video:Lagarith Lossless Codec Audio0:Lautsprecher (2- USB Audio CODEC ) Audio1:Mikrofon (2- USB Audio CODEC )
Duration: 00:27:55.60, start: 0.000000, bitrate: 181780 kb/s
Stream #0:0: Video: lagarith (LAGS / 0x5347414C), rgb24, 1920x1017, 179663 kb/s, 30 fps, 30 tbr, 30 tbn, 30 tbc
Stream #0:1: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, 2 channels, s16, 1411 kb/s
Stream #0:2: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, 1 channels, s16, 705 kb/s
[Parsed_amerge_4 # 000000b1cfaf3200] No channel layout for input 1
Last message repeated 1 times
[AVFilterGraph # 000000b1cfa70c00] The following filters could not choose their formats: Parsed_amerge_4
Consider inserting the (a)format filter near their input or output.
Error configuring complex filters.
Error number -5 occurred

Answer 1

It seems this can be fixed by splitting all audio streams into separate channels first and then feeding it back into the amerge filter.
"ffmpeg.exe"
-i "input.avi"
-filter_complex "
[0:a:0]volume=1.0,channelsplit=channel_layout=stereo[a1][a2];
[0:a:1]volume=0.6,channelsplit=channel_layout=mono[a3];
[a1][a2][a3]amerge=inputs=3,pan=stereo|c0<c0+c2|c1<c1+c2[out]
"
-map 0:v
-map "[out]"
-c:v libx264 -preset slow -crf 26
-c:a aac -strict experimental -ab 128000 -ac 2 -ar 48000
"output.mp4"