[FFmpeg-user] Issue in converting a dss audio file to wav

Wed Jun 9 18:27:28 EEST 2021

Sample file is uploaded here:
https://drive.google.com/file/d/1kwMK-BAityolV9loSPbZESTIXKZowhq5/view?usp=sharing

Output of the conversion:

ffmpeg started on 2021-06-09 at 15:50:57
Report written to "ffmpeg-20210609-155057.log"
Log level: 48
Command line:
ffmpeg -i Welcome.dss output.mp3 -report
ffmpeg version 4.4 Copyright (c) 2000-2021 the FFmpeg developers
built with Apple clang version 12.0.5 (clang-1205.0.22.9)
configuration: --prefix=/opt/homebrew/Cellar/ffmpeg/4.4_1 --enable-shared
--enable-pthreads --enable-version3 --enable-avresample --cc=clang
--host-cflags= --host-ldflags= --enable-ffplay --enable-gnutls --enable-gpl
--enable-libaom --enable-libbluray --enable-libdav1d --enable-libmp3lame
--enable-libopus --enable-librav1e --enable-librubberband
--enable-libsnappy --enable-libsrt --enable-libtesseract --enable-libtheora
--enable-libvidstab --enable-libvorbis --enable-libvpx --enable-libwebp
--enable-libx264 --enable-libx265 --enable-libxml2 --enable-libxvid
--enable-lzma --enable-libfontconfig --enable-libfreetype --enable-frei0r
--enable-libass --enable-libopencore-amrnb --enable-libopencore-amrwb
--enable-libopenjpeg --enable-libspeex --enable-libsoxr --enable-libzmq
--enable-libzimg --disable-libjack --disable-indev=jack
--enable-videotoolbox
libavutil 56. 70.100 / 56. 70.100
libavcodec 58.134.100 / 58.134.100
libavformat 58. 76.100 / 58. 76.100
libavdevice 58. 13.100 / 58. 13.100
libavfilter 7.110.100 / 7.110.100
libavresample 4. 0. 0 / 4. 0. 0
libswscale 5. 9.100 / 5. 9.100
libswresample 3. 9.100 / 3. 9.100
libpostproc 55. 9.100 / 55. 9.100
Splitting the commandline.
Reading option '-i' ... matched as input url with argument 'Welcome.dss'.
Reading option 'output.mp3' ... matched as output url.
Reading option '-report' ... matched as option 'report' (generate a report)
with argument '1'.
Finished splitting the commandline.
Parsing a group of options: global .
Applying option report (generate a report) with argument 1.
Successfully parsed a group of options.
Parsing a group of options: input url Welcome.dss.
Successfully parsed a group of options.
Opening an input file: Welcome.dss.
[NULL @ 0x12101cc00] Opening 'Welcome.dss' for reading
[file @ 0x120f33920] Setting default whitelist 'file,crypto,data'
[dss @ 0x12101cc00] Format dss probed with size=2048 and score=100
[dss @ 0x12101cc00] Before avformat_find_stream_info() pos: 1024 bytes read:
32768 seeks:0 nb_streams:1
[dss @ 0x12101cc00] All info found
[dss @ 0x12101cc00] Estimating duration from bitrate, this may be inaccurate
[dss @ 0x12101cc00] After avformat_find_stream_info() pos: 3104 bytes read:
32768 seeks:0 frames:50
Input #0, dss, from 'Welcome.dss':
Metadata:
author :
date : 2012-01-01T00:00:00
comment :
Duration: 00:00:40.49, start: 0.000000, bitrate: 14 kb/s
Stream #0:0, 50, 1/11025: Audio: dss_sp, 11025 Hz, mono, s16
Successfully opened the file.
Parsing a group of options: output url output.mp3.
Successfully parsed a group of options.
Opening an output file: output.mp3.
[file @ 0x120f34e10] Setting default whitelist 'file,crypto,data'
Successfully opened the file.
Stream mapping:
Stream #0:0 -> #0:0 (dss_sp (native) -> mp3 (libmp3lame))
Press [q] to stop, [?] for help
cur_dts is invalid st:0 (0) [init:0 i_done:0 finish:0] (this is harmless if
it occurs once at the start per stream)
detected 8 logical cores
[graph_0_in_0_0 @ 0x122104240] Setting 'time_base' to value '1/11025'
[graph_0_in_0_0 @ 0x122104240] Setting 'sample_rate' to value '11025'
[graph_0_in_0_0 @ 0x122104240] Setting 'sample_fmt' to value 's16'
[graph_0_in_0_0 @ 0x122104240] Setting 'channel_layout' to value '0x4'
[graph_0_in_0_0 @ 0x122104240] tb:1/11025 samplefmt:s16 samplerate:11025
chlayout:0x4
[format_out_0_0 @ 0x122104730] Setting 'sample_fmts' to value
's32p|fltp|s16p'
[format_out_0_0 @ 0x122104730] Setting 'sample_rates' to value
'44100|48000|32000|22050|24000|16000|11025|12000|8000'
[format_out_0_0 @ 0x122104730] Setting 'channel_layouts' to value '0x4|0x3'
[format_out_0_0 @ 0x122104730] auto-inserting filter 'auto_resampler_0'
between the filter 'Parsed_anull_0' and the filter 'format_out_0_0'
[AVFilterGraph @ 0x1220049d0] query_formats: 4 queried, 6 merged, 3 already
done, 0 delayed
[auto_resampler_0 @ 0x122104ee0] picking s16p out of 3 ref:s16
[auto_resampler_0 @ 0x122104ee0] [SWR @ 0x118008000] Using s16p internally
between filters
[auto_resampler_0 @ 0x122104ee0] ch:1 chl:mono fmt:s16 r:11025Hz -> ch:1
chl:mono fmt:s16p r:11025Hz
Output #0, mp3, to 'output.mp3':
Metadata:
author :
TDRC : 2012-01-01T00:00:00
comment :
TSSE : Lavf58.76.100
Stream #0:0, 0, 1/11025: Audio: mp3, 11025 Hz, mono, s16p
Metadata:
encoder : Lavc58.134.100 libmp3lame
size= 0kB time=00:00:00.00 bitrate=N/A speed= 0x
cur_dts is invalid st:0 (0) [init:1 i_done:0 finish:0] (this is harmless if
it occurs once at the start per stream)
cur_dts is invalid st:0 (0) [init:1 i_done:0 finish:0] (this is harmless if
it occurs once at the start per stream)
cur_dts is invalid st:0 (0) [init:1 i_done:0 finish:0] (this is harmless if
it occurs once at the start per stream)
cur_dts is invalid st:0 (0) [init:1 i_done:0 finish:0] (this is harmless if
it occurs once at the start per stream)
cur_dts is invalid st:0 (0) [init:1 i_done:0 finish:0] (this is harmless if
it occurs once at the start per stream)
cur_dts is invalid st:0 (0) [init:1 i_done:0 finish:0] (this is harmless if
it occurs once at the start per stream)
cur_dts is invalid st:0 (0) [init:1 i_done:0 finish:0] (this is harmless if
it occurs once at the start per stream)
cur_dts is invalid st:0 (0) [init:1 i_done:0 finish:0] (this is harmless if
it occurs once at the start per stream)
cur_dts is invalid st:0 (0) [init:1 i_done:0 finish:0] (this is harmless if
it occurs once at the start per stream)
cur_dts is invalid st:0 (0) [init:1 i_done:0 finish:0] (this is harmless if
it occurs once at the start per stream)
[out_0_0 @ 0x122104500] EOF on sink link out_0_0:default.
No more output streams to write to, finishing.
[libmp3lame @ 0x121029400] Trying to remove 287 more samples than there are
in the queue
size= 80kB time=00:00:40.44 bitrate= 16.1kbits/s speed= 487x
video:0kB audio:79kB subtitle:0kB other streams:0kB global headers:0kB
muxing overhead: 0.397837%
Input file #0 (Welcome.dss):
Input stream #0:0 (audio): 1690 packets read (70980 bytes); 1690 frames
decoded (446160 samples);
Total: 1690 packets (70980 bytes) demuxed
Output file #0 (output.mp3):
Output stream #0:0 (audio): 775 frames encoded (446160 samples); 777
packets muxed (81189 bytes);
Total: 777 packets (81189 bytes) muxed
1690 frames successfully decoded, 0 decoding errors
[AVIOContext @ 0x120f34ed0] Statistics: 0 seeks, 1 writeouts
[AVIOContext @ 0x120f33e30] Statistics: 71168 bytes read, 0 seeks

On Wed, 9 Jun 2021 at 20:52, Carl Zwanzig <cpz at tuunq.com> wrote:

> On 6/9/2021 3:25 AM, Aman Bisht wrote:
> > Please find the attached sample .dss audio used for conversion to .wav.
>
> Please put the file on a sharing site and include the URL.
>
> > I’m using the below command to convert
> > ffmpeg -I Welcome.dss output.wav
> > I have attached the log report.
>
> And please include the complete output in the body of the email.
>
> (As a general rule, if you can't build ffmpeg from the "git head" source,
> which is the only thing actually "supported" here, test with the latest
> nightly build available- see https://ffmpeg.org/mailing-list-faq.html and
> https://ffmpeg.org/download.html)
>
> z!
>
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>
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>

-- 
Thanks & Regards,

*Aman Bisht*
Member of Technical Staff - 1
Antstack Technologies <https://www.antstack.io/>
Contact : +91-9911138685