Dolby E decoder

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Dolby E decoder

Frédéric Busnel-Joncour
Hi there,


First question here !
I've seen that a Dolby E decoder has been added to the last release of libavcodec.

I'm sometimes delivered of stereo 24bit/48Khz wav file that are Dolby E encoded streams.
Is there a way to tell ffplay to play it as Dolby E and decode it ? By default it plays it as a standard PCM wav files.

Sometimes, Dolby E streams are also wrapped into video files, and most of times splitted into two mono audio streams. Is there a way, without any rewrap, to play and decode it through ffplay ? (select both streams > "merge" as a Dolby E stream > decode)

And one last question, is it possible to wrap Dolby-E stream in mov or mxf container using ffmpeg without altering it in any way ?


Best regards,

Fred
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Re: Dolby E decoder

Carl Eugen Hoyos-2
2017-09-07 15:20 GMT+02:00 Frédéric Busnel-Joncour <[hidden email]>:

> I've seen that a Dolby E decoder has been added to the last release of libavcodec.

So far, only SMPTE 337M files are supported, patch for more file types welcome!

Carl Eugen
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Re: Dolby E decoder

Frédéric Busnel-Joncour
Hi Carl Eugen,


Thanks for this quick reply !
Not sure to have well understood: can a SMPTE 337M file be in a wav container, or it has a specific wrap and file extension ?


Best regards,

Fred

> 2017-09-07 15:20 GMT+02:00 Frédéric Busnel-Joncour <[hidden email]>:
>
>> I've seen that a Dolby E decoder has been added to the last release of libavcodec.
>
> So far, only SMPTE 337M files are supported, patch for more file types welcome!
>
> Carl Eugen
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Re: Dolby E decoder

Carl Eugen Hoyos-2
2017-09-07 15:38 GMT+02:00 Frédéric Busnel-Joncour <[hidden email]>:

> Not sure to have well understood: can a SMPTE 337M file be in a wav container, or it has a specific wrap and file extension ?

Afaict, SMPTE 337M is a file format.

Please do not top-post here, Carl Eugen
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Re: Dolby E decoder

Frédéric Busnel-Joncour
In reply to this post by Frédéric Busnel-Joncour

> Afaict, SMPTE 337M is a file format.

OK, thanks. I have made many search but can't find any info about it. With what I've found, it seems to be more like a protocol (non-PCM stream transport in an AES-3 link) than a file format ?

> Please do not top-post here, Carl Eugen

Sorry, I don't understand what you mean ? What should I do to avoid that ?


And What about this question :

> And one last question, is it possible to wrap Dolby-E stream in mov or mxf container using ffmpeg without altering it in any way ?


Best,

Fred


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Re: Dolby E decoder

Carl Eugen Hoyos-2
2017-09-07 18:20 GMT+02:00 Frédéric Busnel-Joncour <[hidden email]>:

>> And one last question, is it possible to wrap Dolby-E stream in mov
>> or mxf container using ffmpeg without altering it in any way ?

Do you have a mov and / or mxf file containing Dolby-E so we
better understand what you request?

(Sorry, my knowledge of SMPTE 337M is limited but I know that so far,
this is the only "format" for which FFmpeg supports Dolby-E, and that
adding support for other formats will be possible but hasn't been
done yet.)

Carl Eugen
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Re: Dolby E decoder

Mike Brown
On Fri, Sep 08, 2017 at 12:54:12AM +0200, Carl Eugen Hoyos wrote:
> (Sorry, my knowledge of SMPTE 337M is limited but I know that so far,
> this is the only "format" for which FFmpeg supports Dolby-E, and that
> adding support for other formats will be possible but hasn't been
> done yet.)

From what I understand, reading the synopsis of SMPTE-337M, is that it is
not a container specifically for Dolby-E.  I couldn't anything out of it to
indicate that it can even handle Dolby-E.

That said, I do deal with SMPTE-302M, which is the spec for which Dolby-E is
applied to Transport Streams.  In my case satellite feeds (not for the
consumer).  The spec talks about MPEG-2 TS streams, but it applies to H.264
streams as well.

Right now, I use ffmpeg to extract the stream into a 2-ch WAVE file, in which
I use in another program to extract the audio into 6 mono WAVE files.  I then
use ffmpeg to encode a AC3 file.

Are you saying that I can use ffmpeg and get to a final ac3 file in a single
step?

MB
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Re: Dolby E decoder

Paul B Mahol
On 9/8/17, Mike Brown <[hidden email]> wrote:

> On Fri, Sep 08, 2017 at 12:54:12AM +0200, Carl Eugen Hoyos wrote:
>> (Sorry, my knowledge of SMPTE 337M is limited but I know that so far,
>> this is the only "format" for which FFmpeg supports Dolby-E, and that
>> adding support for other formats will be possible but hasn't been
>> done yet.)
>
> From what I understand, reading the synopsis of SMPTE-337M, is that it is
> not a container specifically for Dolby-E.  I couldn't anything out of it to
> indicate that it can even handle Dolby-E.
>
> That said, I do deal with SMPTE-302M, which is the spec for which Dolby-E is
> applied to Transport Streams.  In my case satellite feeds (not for the
> consumer).  The spec talks about MPEG-2 TS streams, but it applies to H.264
> streams as well.
>
> Right now, I use ffmpeg to extract the stream into a 2-ch WAVE file, in
> which
> I use in another program to extract the audio into 6 mono WAVE files.  I
> then
> use ffmpeg to encode a AC3 file.
>
> Are you saying that I can use ffmpeg and get to a final ac3 file in a single
> step?

No, you would need to copy audio to raw container, this raw container would be
then detected as Dolby-E.

ffmpeg -i input.wav -c:a copy -f u8 out.u8

ffmpeg -i out.u8 -c:a ac3 .....
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Re: Dolby E decoder

Kieran O Leary
On 8 Sep 2017 08:35, "Paul B Mahol" <[hidden email]> wrote:

On 9/8/17, Mike Brown <

No, you would need to copy audio to raw container, this raw container would
be
then detected as Dolby-E.

ffmpeg -i input.wav -c:a copy -f u8 out.u8


This looks like an 8-bit format,is there a reason why 16 or 24 bit aren't
used in this example, or must u8 be used?
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Re: Dolby E decoder

Paul B Mahol
On 9/8/17, Kieran O Leary <[hidden email]> wrote:

> On 8 Sep 2017 08:35, "Paul B Mahol" <[hidden email]> wrote:
>
> On 9/8/17, Mike Brown <
>
> No, you would need to copy audio to raw container, this raw container would
> be
> then detected as Dolby-E.
>
> ffmpeg -i input.wav -c:a copy -f u8 out.u8
>
>
> This looks like an 8-bit format,is there a reason why 16 or 24 bit aren't
> used in this example, or must u8 be used?

u8 just act as raw muxer, it just copy data from input to output, it
have nothing to do with bit depth.

One can not use s16 or s32 muxer because they require size of packet
multiplied by 2 or 4 which
may not be possible.
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Re: Dolby E decoder

Mike Brown
In reply to this post by Paul B Mahol
On Fri, Sep 08, 2017 at 09:34:50AM +0200, Paul B Mahol wrote:

> On 9/8/17, Mike Brown <[hidden email]> wrote:
> > On Fri, Sep 08, 2017 at 12:54:12AM +0200, Carl Eugen Hoyos wrote:
> >> (Sorry, my knowledge of SMPTE 337M is limited but I know that so far,
> >> this is the only "format" for which FFmpeg supports Dolby-E, and that
> >> adding support for other formats will be possible but hasn't been
> >> done yet.)
> >
> > From what I understand, reading the synopsis of SMPTE-337M, is that it is
> > not a container specifically for Dolby-E.  I couldn't anything out of it to
> > indicate that it can even handle Dolby-E.
> >
> > That said, I do deal with SMPTE-302M, which is the spec for which Dolby-E is
> > applied to Transport Streams.  In my case satellite feeds (not for the
> > consumer).  The spec talks about MPEG-2 TS streams, but it applies to H.264
> > streams as well.
> >
> > Right now, I use ffmpeg to extract the stream into a 2-ch WAVE file, in
> > which
> > I use in another program to extract the audio into 6 mono WAVE files.  I
> > then
> > use ffmpeg to encode a AC3 file.
> >
> > Are you saying that I can use ffmpeg and get to a final ac3 file in a single
> > step?
>
> No, you would need to copy audio to raw container, this raw container would be
> then detected as Dolby-E.
>
> ffmpeg -i input.wav -c:a copy -f u8 out.u8
>
> ffmpeg -i out.u8 -c:a ac3 .....

Could this not be done as a series of pipes:

ffmpeg -i input.ts -map 0:2 -c:a pcm_s24le -f wav - | \
ffmpeg -i - -c:a copy -f u24le - | \
ffmpeg -drc_scale 0 -i - -c:a ac3 -ab 640k -center_mixlev 0.707 output.ac3

Will the 5.1 audio be mapped correctly?

The Dolby-E is 24bit, not 8.

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Re: Dolby E decoder

Paul B Mahol
On 9/8/17, Mike Brown <[hidden email]> wrote:

> On Fri, Sep 08, 2017 at 09:34:50AM +0200, Paul B Mahol wrote:
>> On 9/8/17, Mike Brown <[hidden email]> wrote:
>> > On Fri, Sep 08, 2017 at 12:54:12AM +0200, Carl Eugen Hoyos wrote:
>> >> (Sorry, my knowledge of SMPTE 337M is limited but I know that so far,
>> >> this is the only "format" for which FFmpeg supports Dolby-E, and that
>> >> adding support for other formats will be possible but hasn't been
>> >> done yet.)
>> >
>> > From what I understand, reading the synopsis of SMPTE-337M, is that it
>> > is
>> > not a container specifically for Dolby-E.  I couldn't anything out of it
>> > to
>> > indicate that it can even handle Dolby-E.
>> >
>> > That said, I do deal with SMPTE-302M, which is the spec for which
>> > Dolby-E is
>> > applied to Transport Streams.  In my case satellite feeds (not for the
>> > consumer).  The spec talks about MPEG-2 TS streams, but it applies to
>> > H.264
>> > streams as well.
>> >
>> > Right now, I use ffmpeg to extract the stream into a 2-ch WAVE file, in
>> > which
>> > I use in another program to extract the audio into 6 mono WAVE files.  I
>> > then
>> > use ffmpeg to encode a AC3 file.
>> >
>> > Are you saying that I can use ffmpeg and get to a final ac3 file in a
>> > single
>> > step?
>>
>> No, you would need to copy audio to raw container, this raw container
>> would be
>> then detected as Dolby-E.
>>
>> ffmpeg -i input.wav -c:a copy -f u8 out.u8
>>
>> ffmpeg -i out.u8 -c:a ac3 .....
>
> Could this not be done as a series of pipes:
>
> ffmpeg -i input.ts -map 0:2 -c:a pcm_s24le -f wav - | \
> ffmpeg -i - -c:a copy -f u24le - | \
> ffmpeg -drc_scale 0 -i - -c:a ac3 -ab 640k -center_mixlev 0.707 output.ac3
>
> Will the 5.1 audio be mapped correctly?
>
> The Dolby-E is 24bit, not 8.

Doesn't matter, your example will not work, try it.
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Re: Dolby E decoder

Mike Brown
On Fri, Sep 08, 2017 at 01:15:06PM +0200, Paul B Mahol wrote:

> On 9/8/17, Mike Brown <[hidden email]> wrote:
> > On Fri, Sep 08, 2017 at 09:34:50AM +0200, Paul B Mahol wrote:
> >> On 9/8/17, Mike Brown <[hidden email]> wrote:
> >> > On Fri, Sep 08, 2017 at 12:54:12AM +0200, Carl Eugen Hoyos wrote:
> >> >> (Sorry, my knowledge of SMPTE 337M is limited but I know that so far,
> >> >> this is the only "format" for which FFmpeg supports Dolby-E, and that
> >> >> adding support for other formats will be possible but hasn't been
> >> >> done yet.)
> >> >
> >> > From what I understand, reading the synopsis of SMPTE-337M, is that it
> >> > is
> >> > not a container specifically for Dolby-E.  I couldn't anything out of it
> >> > to
> >> > indicate that it can even handle Dolby-E.
> >> >
> >> > That said, I do deal with SMPTE-302M, which is the spec for which
> >> > Dolby-E is
> >> > applied to Transport Streams.  In my case satellite feeds (not for the
> >> > consumer).  The spec talks about MPEG-2 TS streams, but it applies to
> >> > H.264
> >> > streams as well.
> >> >
> >> > Right now, I use ffmpeg to extract the stream into a 2-ch WAVE file, in
> >> > which
> >> > I use in another program to extract the audio into 6 mono WAVE files.  I
> >> > then
> >> > use ffmpeg to encode a AC3 file.
> >> >
> >> > Are you saying that I can use ffmpeg and get to a final ac3 file in a
> >> > single
> >> > step?
> >>
> >> No, you would need to copy audio to raw container, this raw container
> >> would be
> >> then detected as Dolby-E.
> >>
> >> ffmpeg -i input.wav -c:a copy -f u8 out.u8
> >>
> >> ffmpeg -i out.u8 -c:a ac3 .....
> >
> > Could this not be done as a series of pipes:
> >
> > ffmpeg -i input.ts -map 0:2 -c:a pcm_s24le -f wav - | \
> > ffmpeg -i - -c:a copy -f u24le - | \
> > ffmpeg -drc_scale 0 -i - -c:a ac3 -ab 640k -center_mixlev 0.707 output.ac3
> >
> > Will the 5.1 audio be mapped correctly?
> >
> > The Dolby-E is 24bit, not 8.
>
> Doesn't matter, your example will not work, try it.

Why doesn't it matter?

ffmpeg -i input_test.ts -map 0:1 -c:a pcm_s24le output_test.wav
ffmpeg version N-52837-g399f6ef Copyright (c) 2000-2013 the FFmpeg developers
  built on May  7 2013 01:09:00 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-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. 29.100 / 52. 29.100
  libavcodec     55.  7.100 / 55.  7.100
  libavformat    55.  4.101 / 55.  4.101
  libavdevice    55.  0.100 / 55.  0.100
  libavfilter     3. 63.101 /  3. 63.101
  libswscale      2.  2.100 /  2.  2.100
  libswresample   0. 17.102 /  0. 17.102
  libpostproc    52.  3.100 / 52.  3.100
[mpegts @ 027f7b00] probed stream 4 failed
[mpeg2video @ 027f8600] Invalid frame dimensions 0x0.
    Last message repeated 8 times
[mpegts @ 027f7b00] Could not find codec parameters for stream 4 (Unknown: none (LU-A / 0x412D554C)): unknown codec
Consider increasing the value for the 'analyzeduration' and 'probesize' options
[mpegts @ 027f7b00] PES packet size mismatch
    Last message repeated 2 times
Input #0, mpegts, from 'input_test.ts':
  Duration: 00:00:19.16, start: 94349.992444, bitrate: 43792 kb/s
  Program 1
    Stream #0:0[0x31]: Video: mpeg2video (Main) ([2][0][0][0] / 0x0002), yuv420p, 1920x1080 [SAR 1:1 DAR 16:9], 29.97 fps, 29.97 tbr, 90k tbn, 59.94 tbc
    Stream #0:1[0x33](eng): Audio: s302m (BSSD / 0x44535342), 48000 Hz, stereo, s32, 2304 kb/s
    Stream #0:2[0x34](eng): Audio: mp2 ([3][0][0][0] / 0x0003), 48000 Hz, stereo, s16p, 256 kb/s
    Stream #0:3[0x35](eng): Audio: mp2 ([3][0][0][0] / 0x0003), 48000 Hz, stereo, s16p, 256 kb/s
    Stream #0:4[0x3b]: Unknown: none (LU-A / 0x412D554C)
Output #0, wav, to 'output_test.wav':
  Metadata:
    ISFT            : Lavf55.4.101
    Stream #0:0(eng): Audio: pcm_s24le ([1][0][0][0] / 0x0001), 48000 Hz, stereo, s32, 2304 kb/s
Stream mapping:
  Stream #0:1 -> #0:0 (s302m -> pcm_s24le)
Press [q] to stop, [?] for help
[mpegts @ 027f7b00] PES packet size mismatch
[s302m @ 027f42e0] frame has invalid header
Error while decoding stream #0:1: Invalid data found when processing input
size=    5330kB time=00:00:18.95 bitrate=2304.0kbits/s    
video:0kB audio:5330kB subtitle:0 global headers:0kB muxing overhead 0.001869%


ffmpeg -i output_test.wav -c:a copy -f u24le output_test.u24
ffmpeg version N-52837-g399f6ef Copyright (c) 2000-2013 the FFmpeg developers
  built on May  7 2013 01:09:00 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-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. 29.100 / 52. 29.100
  libavcodec     55.  7.100 / 55.  7.100
  libavformat    55.  4.101 / 55.  4.101
  libavdevice    55.  0.100 / 55.  0.100
  libavfilter     3. 63.101 /  3. 63.101
  libswscale      2.  2.100 /  2.  2.100
  libswresample   0. 17.102 /  0. 17.102
  libpostproc    52.  3.100 / 52.  3.100
[wav @ 027f40a0] max_analyze_duration 5000000 reached at 5001333 microseconds
Input #0, wav, from 'output_test.wav':
  Metadata:
    encoder         : Lavf55.4.101
  Duration: 00:00:18.95, bitrate: 2304 kb/s
    Stream #0:0: Audio: pcm_s24le ([1][0][0][0] / 0x0001), 48000 Hz, stereo, s32, 2304 kb/s
Output #0, u24le, to 'output_test.u24':
  Metadata:
    encoder         : Lavf55.4.101
    Stream #0:0: Audio: pcm_s24le ([1][0][0][0] / 0x0001), 48000 Hz, stereo, 2304 kb/s
Stream mapping:
  Stream #0:0 -> #0:0 (copy)
Press [q] to stop, [?] for help
size=    5330kB time=00:00:18.95 bitrate=2304.0kbits/s    
video:0kB audio:5330kB subtitle:0 global headers:0kB muxing overhead 0.000000%


ffmpeg -drc_scale 0 -i output_test.u24 -c:a ac3 -ab 640k -center_mixlev 0.707 output_test.ac3
ffmpeg version N-52837-g399f6ef Copyright (c) 2000-2013 the FFmpeg developers
  built on May  7 2013 01:09:00 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-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. 29.100 / 52. 29.100
  libavcodec     55.  7.100 / 55.  7.100
  libavformat    55.  4.101 / 55.  4.101
  libavdevice    55.  0.100 / 55.  0.100
  libavfilter     3. 63.101 /  3. 63.101
  libswscale      2.  2.100 /  2.  2.100
  libswresample   0. 17.102 /  0. 17.102
  libpostproc    52.  3.100 / 52.  3.100
output_test.u24: Invalid data found when processing input


When I did it as pipes, the result was: Invalid data found when processing input

So, I did them separately, as shown above.  It made no difference if u8 was
used instead.

MB
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Re: Dolby E decoder

Paul B Mahol
On 9/8/17, Mike Brown <[hidden email]> wrote:

> On Fri, Sep 08, 2017 at 01:15:06PM +0200, Paul B Mahol wrote:
>> On 9/8/17, Mike Brown <[hidden email]> wrote:
>> > On Fri, Sep 08, 2017 at 09:34:50AM +0200, Paul B Mahol wrote:
>> >> On 9/8/17, Mike Brown <[hidden email]> wrote:
>> >> > On Fri, Sep 08, 2017 at 12:54:12AM +0200, Carl Eugen Hoyos wrote:
>> >> >> (Sorry, my knowledge of SMPTE 337M is limited but I know that so
>> >> >> far,
>> >> >> this is the only "format" for which FFmpeg supports Dolby-E, and
>> >> >> that
>> >> >> adding support for other formats will be possible but hasn't been
>> >> >> done yet.)
>> >> >
>> >> > From what I understand, reading the synopsis of SMPTE-337M, is that
>> >> > it
>> >> > is
>> >> > not a container specifically for Dolby-E.  I couldn't anything out of
>> >> > it
>> >> > to
>> >> > indicate that it can even handle Dolby-E.
>> >> >
>> >> > That said, I do deal with SMPTE-302M, which is the spec for which
>> >> > Dolby-E is
>> >> > applied to Transport Streams.  In my case satellite feeds (not for
>> >> > the
>> >> > consumer).  The spec talks about MPEG-2 TS streams, but it applies to
>> >> > H.264
>> >> > streams as well.
>> >> >
>> >> > Right now, I use ffmpeg to extract the stream into a 2-ch WAVE file,
>> >> > in
>> >> > which
>> >> > I use in another program to extract the audio into 6 mono WAVE files.
>> >> >  I
>> >> > then
>> >> > use ffmpeg to encode a AC3 file.
>> >> >
>> >> > Are you saying that I can use ffmpeg and get to a final ac3 file in a
>> >> > single
>> >> > step?
>> >>
>> >> No, you would need to copy audio to raw container, this raw container
>> >> would be
>> >> then detected as Dolby-E.
>> >>
>> >> ffmpeg -i input.wav -c:a copy -f u8 out.u8
>> >>
>> >> ffmpeg -i out.u8 -c:a ac3 .....
>> >
>> > Could this not be done as a series of pipes:
>> >
>> > ffmpeg -i input.ts -map 0:2 -c:a pcm_s24le -f wav - | \
>> > ffmpeg -i - -c:a copy -f u24le - | \
>> > ffmpeg -drc_scale 0 -i - -c:a ac3 -ab 640k -center_mixlev 0.707
>> > output.ac3
>> >
>> > Will the 5.1 audio be mapped correctly?
>> >
>> > The Dolby-E is 24bit, not 8.
>>
>> Doesn't matter, your example will not work, try it.
>
> Why doesn't it matter?
>
> ffmpeg -i input_test.ts -map 0:1 -c:a pcm_s24le output_test.wav

You need -c:a copy here, but you may also need to override input codec.

> ffmpeg version N-52837-g399f6ef Copyright (c) 2000-2013 the FFmpeg
> developers

2013 ? Today year is 2017, Welcome to future.

>   built on May  7 2013 01:09:00 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-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. 29.100 / 52. 29.100
>   libavcodec     55.  7.100 / 55.  7.100
>   libavformat    55.  4.101 / 55.  4.101
>   libavdevice    55.  0.100 / 55.  0.100
>   libavfilter     3. 63.101 /  3. 63.101
>   libswscale      2.  2.100 /  2.  2.100
>   libswresample   0. 17.102 /  0. 17.102
>   libpostproc    52.  3.100 / 52.  3.100
> [mpegts @ 027f7b00] probed stream 4 failed
> [mpeg2video @ 027f8600] Invalid frame dimensions 0x0.
>     Last message repeated 8 times
> [mpegts @ 027f7b00] Could not find codec parameters for stream 4 (Unknown:
> none (LU-A / 0x412D554C)): unknown codec
> Consider increasing the value for the 'analyzeduration' and 'probesize'
> options
> [mpegts @ 027f7b00] PES packet size mismatch
>     Last message repeated 2 times
> Input #0, mpegts, from 'input_test.ts':
>   Duration: 00:00:19.16, start: 94349.992444, bitrate: 43792 kb/s
>   Program 1
>     Stream #0:0[0x31]: Video: mpeg2video (Main) ([2][0][0][0] / 0x0002),
> yuv420p, 1920x1080 [SAR 1:1 DAR 16:9], 29.97 fps, 29.97 tbr, 90k tbn, 59.94
> tbc
>     Stream #0:1[0x33](eng): Audio: s302m (BSSD / 0x44535342), 48000 Hz,
> stereo, s32, 2304 kb/s
>     Stream #0:2[0x34](eng): Audio: mp2 ([3][0][0][0] / 0x0003), 48000 Hz,
> stereo, s16p, 256 kb/s
>     Stream #0:3[0x35](eng): Audio: mp2 ([3][0][0][0] / 0x0003), 48000 Hz,
> stereo, s16p, 256 kb/s
>     Stream #0:4[0x3b]: Unknown: none (LU-A / 0x412D554C)
> Output #0, wav, to 'output_test.wav':
>   Metadata:
>     ISFT            : Lavf55.4.101
>     Stream #0:0(eng): Audio: pcm_s24le ([1][0][0][0] / 0x0001), 48000 Hz,
> stereo, s32, 2304 kb/s
> Stream mapping:
>   Stream #0:1 -> #0:0 (s302m -> pcm_s24le)
> Press [q] to stop, [?] for help
> [mpegts @ 027f7b00] PES packet size mismatch
> [s302m @ 027f42e0] frame has invalid header
> Error while decoding stream #0:1: Invalid data found when processing input
> size=    5330kB time=00:00:18.95 bitrate=2304.0kbits/s
> video:0kB audio:5330kB subtitle:0 global headers:0kB muxing overhead
> 0.001869%
>
>
> ffmpeg -i output_test.wav -c:a copy -f u24le output_test.u24

Too late for -c:a copy.


> ffmpeg version N-52837-g399f6ef Copyright (c) 2000-2013 the FFmpeg
> developers
>   built on May  7 2013 01:09:00 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-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. 29.100 / 52. 29.100
>   libavcodec     55.  7.100 / 55.  7.100
>   libavformat    55.  4.101 / 55.  4.101
>   libavdevice    55.  0.100 / 55.  0.100
>   libavfilter     3. 63.101 /  3. 63.101
>   libswscale      2.  2.100 /  2.  2.100
>   libswresample   0. 17.102 /  0. 17.102
>   libpostproc    52.  3.100 / 52.  3.100
> [wav @ 027f40a0] max_analyze_duration 5000000 reached at 5001333
> microseconds
> Input #0, wav, from 'output_test.wav':
>   Metadata:
>     encoder         : Lavf55.4.101
>   Duration: 00:00:18.95, bitrate: 2304 kb/s
>     Stream #0:0: Audio: pcm_s24le ([1][0][0][0] / 0x0001), 48000 Hz, stereo,
> s32, 2304 kb/s
> Output #0, u24le, to 'output_test.u24':
>   Metadata:
>     encoder         : Lavf55.4.101
>     Stream #0:0: Audio: pcm_s24le ([1][0][0][0] / 0x0001), 48000 Hz, stereo,
> 2304 kb/s
> Stream mapping:
>   Stream #0:0 -> #0:0 (copy)
> Press [q] to stop, [?] for help
> size=    5330kB time=00:00:18.95 bitrate=2304.0kbits/s
> video:0kB audio:5330kB subtitle:0 global headers:0kB muxing overhead
> 0.000000%
>
>
> ffmpeg -drc_scale 0 -i output_test.u24 -c:a ac3 -ab 640k -center_mixlev
> 0.707 output_test.ac3
> ffmpeg version N-52837-g399f6ef Copyright (c) 2000-2013 the FFmpeg
> developers
>   built on May  7 2013 01:09:00 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-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. 29.100 / 52. 29.100
>   libavcodec     55.  7.100 / 55.  7.100
>   libavformat    55.  4.101 / 55.  4.101
>   libavdevice    55.  0.100 / 55.  0.100
>   libavfilter     3. 63.101 /  3. 63.101
>   libswscale      2.  2.100 /  2.  2.100
>   libswresample   0. 17.102 /  0. 17.102
>   libpostproc    52.  3.100 / 52.  3.100
> output_test.u24: Invalid data found when processing input
>
>
> When I did it as pipes, the result was: Invalid data found when processing
> input
>
> So, I did them separately, as shown above.  It made no difference if u8 was
> used instead.
>
> MB
> --
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> Sherlock - The Abominable Bride - 1/01/16
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Re: Dolby E decoder

Cley Faye
In reply to this post by Mike Brown
>
>
> > >> ffmpeg -i input.wav -c:a copy -f u8 out.u8
>


> > > ffmpeg -i - -c:a copy -f u24le - | \
>


> > > The Dolby-E is 24bit, not 8.
> >
> > Doesn't matter, your example will not work, try it.
>
> Why doesn't it matter?
>
>
​When you copy a stream as-is, copying each byte individually or grouping
them by 3 doesn't matter. The "copy"​ codec simply pass the bytes making up
the stream without interpretation. Since all streams are made of bytes (at
least when put in a file), -c copy -f u8 will always work, whatever the
actual stream might be.

​This is just about the copy codec; why the remainder fail I have no idea.​
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Re: Dolby E decoder

Mike Brown
In reply to this post by Paul B Mahol
On Fri, Sep 08, 2017 at 01:39:34PM +0200, Paul B Mahol wrote:
> On 9/8/17, Mike Brown <[hidden email]> wrote:

[trimmed]

> > Why doesn't it matter?
> >
> > ffmpeg -i input_test.ts -map 0:1 -c:a pcm_s24le output_test.wav
>
> You need -c:a copy here, but you may also need to override input codec.

The initial example wanted a wave file.  The audio stream that is in the
TS file is not wave, it is SMPTE-302M.  The above extracts the wave stream
out of the TS file's PCM stream.  Obviously I am missing something as to
what is going on.

So, I am completely confused as to how the raw stream data can be extracted,
or overridden.

> > ffmpeg version N-52837-g399f6ef Copyright (c) 2000-2013 the FFmpeg
> > developers
>
> 2013 ? Today year is 2017, Welcome to future.

That is because newer versions of ffmpeg no longer work with XP.  They crash
trying to use an operation in a DLL that does not exist.  I have operational
reasons that require XP.

MB
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Re: Dolby E decoder

Paul B Mahol
On 9/8/17, Mike Brown <[hidden email]> wrote:

> On Fri, Sep 08, 2017 at 01:39:34PM +0200, Paul B Mahol wrote:
>> On 9/8/17, Mike Brown <[hidden email]> wrote:
>
> [trimmed]
>
>> > Why doesn't it matter?
>> >
>> > ffmpeg -i input_test.ts -map 0:1 -c:a pcm_s24le output_test.wav
>>
>> You need -c:a copy here, but you may also need to override input codec.
>
> The initial example wanted a wave file.  The audio stream that is in the
> TS file is not wave, it is SMPTE-302M.  The above extracts the wave stream
> out of the TS file's PCM stream.  Obviously I am missing something as to
> what is going on.
>
> So, I am completely confused as to how the raw stream data can be extracted,
> or overridden.
>
>> > ffmpeg version N-52837-g399f6ef Copyright (c) 2000-2013 the FFmpeg
>> > developers
>>
>> 2013 ? Today year is 2017, Welcome to future.
>
> That is because newer versions of ffmpeg no longer work with XP.  They crash
> trying to use an operation in a DLL that does not exist.  I have operational
> reasons that require XP.

Than no help for you here.
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Re: Dolby E decoder

Mike Brown
On Fri, Sep 08, 2017 at 02:30:08PM +0200, Paul B Mahol wrote:

> On 9/8/17, Mike Brown <[hidden email]> wrote:
> > On Fri, Sep 08, 2017 at 01:39:34PM +0200, Paul B Mahol wrote:
> >> On 9/8/17, Mike Brown <[hidden email]> wrote:
> >
> > [trimmed]
> >
> >> > Why doesn't it matter?
> >> >
> >> > ffmpeg -i input_test.ts -map 0:1 -c:a pcm_s24le output_test.wav
> >>
> >> You need -c:a copy here, but you may also need to override input codec.
> >
> > The initial example wanted a wave file.  The audio stream that is in the
> > TS file is not wave, it is SMPTE-302M.  The above extracts the wave stream
> > out of the TS file's PCM stream.  Obviously I am missing something as to
> > what is going on.
> >
> > So, I am completely confused as to how the raw stream data can be extracted,
> > or overridden.
> >
> >> > ffmpeg version N-52837-g399f6ef Copyright (c) 2000-2013 the FFmpeg
> >> > developers
> >>
> >> 2013 ? Today year is 2017, Welcome to future.
> >
> > That is because newer versions of ffmpeg no longer work with XP.  They crash
> > trying to use an operation in a DLL that does not exist.  I have operational
> > reasons that require XP.
>
> Than no help for you here.

Are you saying that it takes a version newer than what I am using in order
to decode Dolby-E to ac3?

I have a Win7 box that I recently brought up and can do some testing with
a later version of ffmpeg.  In this case, what do I need to do for this
override you are talking about?

MB
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Re: Dolby E decoder

Paul B Mahol
On 9/8/17, Mike Brown <[hidden email]> wrote:

> On Fri, Sep 08, 2017 at 02:30:08PM +0200, Paul B Mahol wrote:
>> On 9/8/17, Mike Brown <[hidden email]> wrote:
>> > On Fri, Sep 08, 2017 at 01:39:34PM +0200, Paul B Mahol wrote:
>> >> On 9/8/17, Mike Brown <[hidden email]> wrote:
>> >
>> > [trimmed]
>> >
>> >> > Why doesn't it matter?
>> >> >
>> >> > ffmpeg -i input_test.ts -map 0:1 -c:a pcm_s24le output_test.wav
>> >>
>> >> You need -c:a copy here, but you may also need to override input codec.
>> >
>> > The initial example wanted a wave file.  The audio stream that is in the
>> > TS file is not wave, it is SMPTE-302M.  The above extracts the wave
>> > stream
>> > out of the TS file's PCM stream.  Obviously I am missing something as to
>> > what is going on.
>> >
>> > So, I am completely confused as to how the raw stream data can be
>> > extracted,
>> > or overridden.
>> >
>> >> > ffmpeg version N-52837-g399f6ef Copyright (c) 2000-2013 the FFmpeg
>> >> > developers
>> >>
>> >> 2013 ? Today year is 2017, Welcome to future.
>> >
>> > That is because newer versions of ffmpeg no longer work with XP.  They
>> > crash
>> > trying to use an operation in a DLL that does not exist.  I have
>> > operational
>> > reasons that require XP.
>>
>> Than no help for you here.
>
> Are you saying that it takes a version newer than what I am using in order
> to decode Dolby-E to ac3?

Yes.

>
> I have a Win7 box that I recently brought up and can do some testing with
> a later version of ffmpeg.  In this case, what do I need to do for this
> override you are talking about?

Override is doable only with wav files i think.
TS files are not possible, last time I tried.
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Re: Dolby E decoder

Mike Brown
On Fri, Sep 08, 2017 at 03:13:46PM +0200, Paul B Mahol wrote:
> On 9/8/17, Mike Brown <[hidden email]> wrote:
> > Are you saying that it takes a version newer than what I am using in order
> > to decode Dolby-E to ac3?
>
> Yes.

Well, I have the latest available from the website installed on the Win7
box and the test file copied over as well.

> > I have a Win7 box that I recently brought up and can do some testing with
> > a later version of ffmpeg.  In this case, what do I need to do for this
> > override you are talking about?
>
> Override is doable only with wav files i think.
> TS files are not possible, last time I tried.

That is where the -map is used to only deal with the PCM audio stream, but:

ffmpeg -i input_test.ts -map 0:1 -c:a copy -f u8 output_test.u8
ffmpeg version N-87196-g6cadbb1 Copyright (c) 2000-2017 the FFmpeg developers
  built with gcc 7.1.0 (GCC)
  configuration: --enable-gpl --enable-version3 --enable-cuda --enable-cuvid --enable-d3d11va --enable-dxva2 --enable-libmfx --enable-nvenc --enable-avisynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enable-iconv --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libfreetype --enable-libgme --enable-libgsm --enable-libilbc --enable-libmodplug --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenh264 --enable-libopenjpeg --enable-libopus --enable-librtmp --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxavs --enable-libxvid --enable-libzimg --enable-lzma --enable-zlib
  libavutil      55. 74.100 / 55. 74.100
  libavcodec     57.105.100 / 57.105.100
  libavformat    57. 81.100 / 57. 81.100
  libavdevice    57.  8.100 / 57.  8.100
  libavfilter     6.101.100 /  6.101.100
  libswscale      4.  7.103 /  4.  7.103
  libswresample   2.  8.100 /  2.  8.100
  libpostproc    54.  6.100 / 54.  6.100
[s302m @ 000000000051d420] S302 non PCM mode with data type 28 not supported
    Last message repeated 1 times
[mpeg2video @ 000000000051c4e0] Invalid frame dimensions 0x0.
[s302m @ 000000000051d420] S302 non PCM mode with data type 28 not supported
[mpeg2video @ 000000000051c4e0] Invalid frame dimensions 0x0.
[s302m @ 000000000051d420] S302 non PCM mode with data type 28 not supported
[mpeg2video @ 000000000051c4e0] Invalid frame dimensions 0x0.
[s302m @ 000000000051d420] S302 non PCM mode with data type 28 not supported
[mpeg2video @ 000000000051c4e0] Invalid frame dimensions 0x0.
[s302m @ 000000000051d420] S302 non PCM mode with data type 28 not supported
[mpeg2video @ 000000000051c4e0] Invalid frame dimensions 0x0.
[s302m @ 000000000051d420] S302 non PCM mode with data type 28 not supported
[mpeg2video @ 000000000051c4e0] Invalid frame dimensions 0x0.
[s302m @ 000000000051d420] S302 non PCM mode with data type 28 not supported
[mpeg2video @ 000000000051c4e0] Invalid frame dimensions 0x0.
[s302m @ 000000000051d420] S302 non PCM mode with data type 28 not supported
[mpeg2video @ 000000000051c4e0] Invalid frame dimensions 0x0.
[s302m @ 000000000051d420] S302 non PCM mode with data type 28 not supported
[mpeg2video @ 000000000051c4e0] Invalid frame dimensions 0x0.
[s302m @ 000000000051d420] S302 non PCM mode with data type 28 not supported
    Last message repeated 18 times
[mpegts @ 000000000051a9c0] decoding for stream 1 failed
[mpegts @ 000000000051a9c0] PES packet size mismatch
    Last message repeated 2 times
[mpegts @ 000000000051a9c0] Could not find codec parameters for stream 1 (Audio: s302m (BSSD / 0x44535342), stereo, s32 (20 bit), 2304 kb/s): unspecified sample rate
Consider increasing the value for the 'analyzeduration' and 'probesize' options
Input #0, mpegts, from 'input_test.ts':
  Duration: 00:00:19.19, start: 94349.992444, bitrate: 43715 kb/s
  Program 1
    Stream #0:0[0x31]: Video: mpeg2video (Main) ([2][0][0][0] / 0x0002), yuv420p(tv, bt709, top first), 1920x1080 [SAR 1:1 DAR 16:9], Closed Captions, 40266 kb/s, 29.97 fps, 29.97 tbr, 90k tbn, 59.94 tbc
    Stream #0:1[0x33](eng): Audio: s302m (BSSD / 0x44535342), stereo, s32 (20 bit), 2304 kb/s
    Stream #0:2[0x34](eng): Audio: mp2 ([3][0][0][0] / 0x0003), 48000 Hz, stereo, s16p, 256 kb/s
    Stream #0:3[0x35](eng): Audio: mp2 ([3][0][0][0] / 0x0003), 48000 Hz, stereo, s16p, 256 kb/s
    Stream #0:4[0x3b]: Data: bin_data (LU-A / 0x412D554C)
[u8 @ 0000000002330f80] sample rate not set
Could not write header for output file #0 (incorrect codec parameters ?): Invalid argument
Stream mapping:
  Stream #0:1 -> #0:0 (copy)
    Last message repeated 1 times


I also tried to run the command that I normally use to extract the wav file
from the PCM stream and I got the above errors.  The newest ffmpeg has been
broken.  I remember this issue.  I think I found a newer version that ran
under XP, but it had the above errors.  There is history of this in the archive.
So, I had to go back to a version where ffmpeg would extract the PCM data
and place it into the wav file.

MB
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