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

2021-06-10 Thread Aman Bisht
Hi Carl,

I built and installed ffmpeg from git head but the issue is still there.

ffmpeg started on 2021-06-10 at 12:48:51
Report written to "ffmpeg-20210610-124851.log"
Log level: 48
Command line:
ffmpeg -i Welcome.dss -acodec pcm_s16le output.wav -report
ffmpeg version N-102686-g99bbf4a5be Copyright (c) 2000-2021 the FFmpeg 
developers
  built with Apple clang version 12.0.5 (clang-1205.0.22.9)
  configuration: 
  libavutil  57.  0.100 / 57.  0.100
  libavcodec 59.  1.101 / 59.  1.101
  libavformat59.  2.102 / 59.  2.102
  libavdevice59.  0.100 / 59.  0.100
  libavfilter 8.  0.101 /  8.  0.101
  libswscale  6.  0.100 /  6.  0.100
  libswresample   4.  0.100 /  4.  0.100
Splitting the commandline.
Reading option '-i' ... matched as input url with argument 'Welcome.dss'.
Reading option '-acodec' ... matched as option 'acodec' (force audio codec 
('copy' to copy stream)) with argument 'pcm_s16le'.
Reading option 'output.wav' ... 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 @ 0x137e190d0] Opening 'Welcome.dss' for reading
[file @ 0x137e19470] Setting default whitelist 'file,crypto,data'
[dss @ 0x137e190d0] Format dss probed with size=2048 and score=100
[dss @ 0x137e190d0] Before avformat_find_stream_info() pos: 1024 bytes 
read:32768 seeks:0 nb_streams:1
[dss @ 0x137e190d0] All info found
[dss @ 0x137e190d0] Estimating duration from bitrate, this may be inaccurate
[dss @ 0x137e190d0] 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: 0.00, 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.wav.
Applying option acodec (force audio codec ('copy' to copy stream)) with 
argument pcm_s16le.
Successfully parsed a group of options.
Opening an output file: output.wav.
[file @ 0x127e04610] Setting default whitelist 'file,crypto,data'
Successfully opened the file.
Stream mapping:
  Stream #0:0 -> #0:0 (dss_sp (native) -> pcm_s16le (native))
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 @ 0x137f0de50] Setting 'time_base' to value '1/11025'
[graph_0_in_0_0 @ 0x137f0de50] Setting 'sample_rate' to value '11025'
[graph_0_in_0_0 @ 0x137f0de50] Setting 'sample_fmt' to value 's16'
[graph_0_in_0_0 @ 0x137f0de50] Setting 'channel_layout' to value '0x4'
[graph_0_in_0_0 @ 0x137f0de50] tb:1/11025 samplefmt:s16 samplerate:11025 
chlayout:0x4
[format_out_0_0 @ 0x137f0fba0] Setting 'sample_fmts' to value 's16'
[AVFilterGraph @ 0x127e057d0] query_formats: 4 queried, 9 merged, 0 already 
done, 0 delayed
Output #0, wav, to 'output.wav':
  Metadata:
author  : 
ICRD: 2012-01-01T00:00:00
ICMT: 
ISFT: Lavf59.2.102
  Stream #0:0, 0, 1/11025: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 11025 Hz, 
mono, s16, 176 kb/s
Metadata:
  encoder : Lavc59.1.101 pcm_s16le
size=   1kB time=00:00:00.02 bitrate= 211.8kbits/s speed=1.84e+03x
[out_0_0 @ 0x137f0ffb0] EOF on sink link out_0_0:default.
No more output streams to write to, finishing.
size= 872kB time=00:00:40.46 bitrate= 176.4kbits/s speed=1.77e+03x
video:0kB audio:871kB subtitle:0kB other streams:0kB global headers:0kB muxing 
overhead: 0.011879%
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.wav):
  Output stream #0:0 (audio): 1690 frames encoded (446160 samples); 1690 
packets muxed (892320 bytes); 
  Total: 1690 packets (892320 bytes) muxed
1690 frames successfully decoded, 0 decoding errors
[AVIOContext @ 0x127e04680] Statistics: 4 seeks, 6 writeouts
[AVIOContext @ 0x137e19580] Statistics: 71168 bytes read, 0 seeks



> On 10-Jun-2021, at 2:26 AM, Carl Eugen Hoyos  wrote:
> 
> Am Mi., 9. Juni 2021 um 18:00 Uhr schrieb Aman Bisht :
> 
>> ffmpeg version 4.4 Copyright (c) 2000-2021 the FFmpeg developers
> 
> Is the issue you see reproducible with current FFmpeg git head?
> 
> Carl Eugen
> ___
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> ffmpeg-user@ffmpeg.org
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> To unsubscr

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

2021-06-09 Thread Aman Bisht
 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  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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> ffmpeg-user@ffmpeg.org
> https://ffmpeg.org/mailman/listinfo/ffmpeg-user
>
> To unsubscribe, visit link above, or email
> ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
>


-- 
Thanks & Regards,

*Aman Bisht*
Member of Technical Staff - 1
Antstack Technologies <https://www.antstack.io/>
Contact : +91-9911138685
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[FFmpeg-user] Issue in converting a dss audio file to wav

2021-06-09 Thread Aman Bisht
Hi team,

I am using ffmpeg for a use case and found a strange thing in the audio 
conversion process. I haven’t been able to find related material/docs that I 
could use to resolve the issue.

Please find the attached sample .dss audio used for conversion to .wav. I’m 
using the below command to convert

ffmpeg -I Welcome.dss output.wav

I have attached the log report. While the conversion does happen a strange 
noise has been added at the end to the converted audio. 

I would really appreciate if you could assist me resolving this issue and guide 
me to related docs so that I could resolve this.

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