AVANTI - FFmpeg/Avisynth GUI (support thread).

[Chris K]

Wanted to know if I have a watchfolder which i am using for creating .mxf files, can I use same watchfolder to create same .mxf output but using stitch option. Means I have to use one watchfolder for single source and for stitching source files.

Don't know exactly what you mean with stitching?

When the watch option is disabled (no red box on the left), you can use both the source (watch folder) and destination folders for normal conversions. Clips that are written to the watch folder when it is disabled are not seen if you enable the watch option again. The watch option is only triggered on newly copied clips when it is enabled. You cannot enable the watch option and process clips manually at the same time.

Chris

[GoldenMeanie]

Okay I've got another problem I hope can be fixed. I know that FFMPEG will only join VOB files and one other type of container (can't remember) but I think what I really want to do is "Re-mux" and I'm still shakey on the whole muxing thing. My source files are also a bit weird. I render animated clips using an old 3D program called Cool 3D from Ulead systems and there is no audio stream in the final animation. I create simple lead ins for my video that need to be joined at the beginning of my main edit. So basically I need to join two MP4 video files, with the first one having no audio stream. How can this be done?

What you want, join/remux, is not the easiest thing to do!

Both the lead-in and the main clip need to have the exact same audio and video settings.

Assuming your lead-in is already in the proper video format, load the clip at "Source 1", set the audio settings at the main page the same as your main clip and the video settings to "Copy Video". Then copy the following to the FFmpeg script editor...

[USER_VIDEO_OPTIONS]

-new

-i "@source1"
-f lavfi -i "aevalsrc=0"
-shortest
-dvs
-das
-o "@destin1"

This will remux the lead-in video and adds a silent audio stream with the proper settings.

Then use the concat demuxer (see help) to join/remux the 2 clips (both audio and video set to copy).

Thank you so much. Before I read your script I actually found a workaround using AVIDemux but realized (as you have pointed out) I had to add an audio stream to the original silent clip. Easy enough to do using Avanti and Adobe Audition. I created a silent WAV file and then added it to the file while copying the video. But this leads me to ask you another question:

I was under the impression that if you had an audio file placed in Source 2, it would replace the audio file in source 1, provided it's the exact same length. I have found however that it appears as though they will be mixed together. Is this right?

I was having terrible noise on my final video to the point where I had to strip the audio from the master files, mix my music and audio with Audition as a WAV file, then saved back to the silent master using Avanti. Just wondering if I assumed wrong.

[GoldenMeanie]

Hi, it's me again. Got another question if you don't mind:

Sometimes while converting I get repeated VBV UNDERFLOW errors on my frames when converting. After about 28 warnings it says "consider to cancel process" but it converts okay and the video plays fine. I'm wondering what does this mean and how do I avoid it in the future?

[Chris K]

I was under the impression that if you had an audio file placed in Source 2, it would replace the audio file in source 1, provided it's the exact same length. I have found however that it appears as though they will be mixed together. Is this right?

No, the audio clip at "Source 2" replaces the audio from the clip at "Source 1". You should be able to check that out by playing the clip in a media player like e.g. WMP.

The clip at "Source 1" may also be a video only clip. If the audio duration at "Source 2" is longer than the duration of the "Source 1" video, you'll get a message box with the option to truncate the audio.

Sometimes while converting I get repeated VBV UNDERFLOW errors on my frames when converting. After about 28 warnings it says "consider to cancel process" but it converts okay and the video plays fine. I'm wondering what does this mean and how do I avoid it in the future?

You should not have that message. Even if it plays fine on your PC, you could get issues on a stand alone player or TV player.

I don't know what your settings are but if there is a value set in the "VBV buffer size" field at the main page, you'd probably have to increase that value.

Chris

[GoldenMeanie]

aaaand another question....

This one is really tripping me up.

I'm trying to convert an H.264 MOV file (here's the details)

Duration: 00:00:09.01, start: 0.000000, bitrate: 13143 kb/s
Stream #0:0(eng): Video: h264 (Main) (avc1 / 0x31637661), yuv420p, 1280x720,
12286 kb/s, 59.94 fps, 59.94 tbr, 60k tbn, 120k tbc (default)
Stream #0:1(eng): Audio: pcm_s16le (sowt / 0x74776F73), 48000 Hz, mono, s16,
768 kb/s (default)

to MP4. I would just copy the video but I need to downscale it from 59.94fps to 29.97fps.

Output #0, mp4, to 'C:/Documents and Settings/Owner/Desktop/DownsizeTest.mp4':
Stream #0:0(eng): Video: h264 (libx264) ([33][0][0][0] / 0x0021), yuv420p,
1280x720 [SAR 1:1 DAR 16:9], q=-1-10, 13009 kb/s, 11988 tbn,
29.97 tbc (default)
Stream #0:1(eng): Audio: aac (libvo_aacenc) ([64][0][0][0] / 0x0040),
44100 Hz, mono, s16, 320 kb/s (default)

The problem I'm running into is the filesize is over twice the size of the original.
Even when I use the calculator it says it should be about the same (14mb) but I get a file that's 36mb.

What
The
Hell?



ADDENDUM:

I have since tried processing using no variable bitrate options and I did not get the VBV error so that much is solved.
However I'm still getting an ungodly huge filesize and I've noticed it's compressing to a bitrate of 32,000kbs instead of 14000kbs as my settings require. Hence that's why my filesizes are so huge but why would it DOUBLE the bitrate? I'm checking my settings, not sure what I'm doing wrong yet.

[GoldenMeanie]

I was under the impression that if you had an audio file placed in Source 2, it would replace the audio file in source 1, provided it's the exact same length. I have found however that it appears as though they will be mixed together. Is this right?

No, the audio clip at "Source 2" replaces the audio from the clip at "Source 1". You should be able to check that out by playing the clip in a media player like e.g. WMP.

The clip at "Source 1" may also be a video only clip. If the audio duration at "Source 2" is longer than the duration of the "Source 1" video, you'll get a message box with the option to truncate the audio.

Chris

Yes I've seen that before, nonetheless when I used the clip with the original audio it sounded as if the system was trying to combine the two and ambient background noise had that "warbling" that's associated with bad audio compression. Not until I actually separated the audio from the video, then re-combined with the master audio with the silent video did it sound okay.

I don't know what your settings are but if there is a value set in the "VBV buffer size" field at the main page, you'd probably have to increase that value.

Chris

That seems to help but I still get the warning and I've increased the buffer to some ungodly value. Is there a limit to the buffer size?
Sorry for being so stupid.

[Chris K]

aaaand another question....

This one is really tripping me up.

I'm trying to convert an H.264 MOV file (here's the details)

Duration: 00:00:09.01, start: 0.000000, bitrate: 13143 kb/s
Stream #0:0(eng): Video: h264 (Main) (avc1 / 0x31637661), yuv420p, 1280x720,
12286 kb/s, 59.94 fps, 59.94 tbr, 60k tbn, 120k tbc (default)
Stream #0:1(eng): Audio: pcm_s16le (sowt / 0x74776F73), 48000 Hz, mono, s16,
768 kb/s (default)

to MP4. I would just copy the video but I need to downscale it from 59.94fps to 29.97fps.

Output #0, mp4, to 'C:/Documents and Settings/Owner/Desktop/DownsizeTest.mp4':
Stream #0:0(eng): Video: h264 (libx264) ([33][0][0][0] / 0x0021), yuv420p,
1280x720 [SAR 1:1 DAR 16:9], q=-1-10, 13009 kb/s, 11988 tbn,
29.97 tbc (default)
Stream #0:1(eng): Audio: aac (libvo_aacenc) ([64][0][0][0] / 0x0040),
44100 Hz, mono, s16, 320 kb/s (default)

The problem I'm running into is the filesize is over twice the size of the original.
Even when I use the calculator it says it should be about the same (14mb) but I get a file that's 36mb.

I assume the source mov is interlaced (59.94 fps)?

In the output I see q=-1-10. Do you use -qmax 10 for x264? Common values are -qmin 10 -qmax 51 or leave that off and try -crf 19.

I'm not really a x264 expert where it concerns all the available settings.

[Chris K]

That seems to help but I still get the warning and I've increased the buffer to some ungodly value. Is there a limit to the buffer size?

I've looked to one of my saved templates (not in the release). With 1280x720 frame size and this settings on the main page...

Bitrate 14000
Max rate 14000
VBV buffer size 30000

at the FFmpeg script editor, I have this...

[USER_VIDEO_OPTIONS]

-preset superfast
-tune film
-bluray-compat 1
-keyint_min 60
-level 41

I don't get any buffer underflows and bitrte varies around 14000.

P.S., It's past 2 AM here in Europe. I'll have to go off line for a little nap. We'll discus the subject later!

[GoldenMeanie]

aaaand another question....

This one is really tripping me up.

I'm trying to convert an H.264 MOV file (here's the details)

Duration: 00:00:09.01, start: 0.000000, bitrate: 13143 kb/s
Stream #0:0(eng): Video: h264 (Main) (avc1 / 0x31637661), yuv420p, 1280x720,
12286 kb/s, 59.94 fps, 59.94 tbr, 60k tbn, 120k tbc (default)
Stream #0:1(eng): Audio: pcm_s16le (sowt / 0x74776F73), 48000 Hz, mono, s16,
768 kb/s (default)

to MP4. I would just copy the video but I need to downscale it from 59.94fps to 29.97fps.

Output #0, mp4, to 'C:/Documents and Settings/Owner/Desktop/DownsizeTest.mp4':
Stream #0:0(eng): Video: h264 (libx264) ([33][0][0][0] / 0x0021), yuv420p,
1280x720 [SAR 1:1 DAR 16:9], q=-1-10, 13009 kb/s, 11988 tbn,
29.97 tbc (default)
Stream #0:1(eng): Audio: aac (libvo_aacenc) ([64][0][0][0] / 0x0040),
44100 Hz, mono, s16, 320 kb/s (default)

The problem I'm running into is the filesize is over twice the size of the original.
Even when I use the calculator it says it should be about the same (14mb) but I get a file that's 36mb.

I assume the source mov is interlaced (59.94 fps)?

In the output I see q=-1-10. Do you use -qmax 10 for x264? Common values are -qmin 10 -qmax 51 or leave that off and try -crf 19.

I'm not really a x264 expert where it concerns all the available settings.

Since you've gone to bed I won't expect a reply till later if at all, but here's what I've done since our last correspondence:

I deleted the INI file so I could start fresh.
I removed all Variable Bitrate settings and set it for 14000kbs. All other settings are the same.

I'm not sure about the video being interlaced but the process worked, yet it literally took an hour to process nine seconds of video.
The result was what I wanted, but an hour seems way too excessive to me.
Any clues?

I don't usually adjust any qmax settings. I barely know how to calculate bitrates.

[Chris K]

I'm not sure about the video being interlaced but the process worked, yet it literally took an hour to process nine seconds of video.
The result was what I wanted, but an hour seems way too excessive to me.
Any clues?

There must be something outside Avanti that awefully slows down your system. I would first try to find out and fix this before you continue.

I don't have a source with your specs available but to illustrate I did the following.

1. Selected the H.264 option at the codec field (loads a default x264 template).
2. Set video bitrate to 14000 (left all other bitrate fields empty).
3. Set frame size to 1280x720 (as to your needs).
4. Changed the AAC settings as to your needs (320kbs, 44100hz, 1 channel).

My source is a PAL HD clip of 12 secs. I run this on my old AMD Athlon machine which should be slower than a Pentium 4. With this I get the following output which took slightly more than four minutes.

7:47:50 - Process started using ffmpeg version N-69406-ga73c411
======================================================================================

Input #0, mpegts, from 'F:/x-test_videos/00081.mts':
  Duration: 00:00:12.51, start: 0.541767, bitrate: 12786 kb/s
  Program 1 
    Stream #0:0[0x1011]: Video: h264 (High) (HDMV / 0x564D4448), yuv420p,
      1920x1080 [SAR 1:1 DAR 16:9], 25 fps, 25 tbr, 90k tbn, 50 tbc
    Stream #0:1[0x1100]: Audio: ac3 (AC-3 / 0x332D4341), 48000 Hz, stereo, fltp,
      256 kb/s
    Stream #0:2[0x1200]: Subtitle: hdmv_pgs_subtitle ([144][0][0][0] / 0x0090),
      1920x1080
---------------------------------------------------------------------------------
Output #0, mp4, to 'F:/VirtualDub Capture2/avi_test_final.mp4':
    Stream #0:0: Video: h264 (libx264) ([33][0][0][0] / 0x0021), yuv420p,
      1280x720 [SAR 1:1 DAR 16:9], q=10-51, 14000 kb/s, 25 fps, 12800 tbn,
      25 tbc
    Stream #0:1: Audio: aac (libvo_aacenc) ([64][0][0][0] / 0x0040), 44100 Hz,
      mono, s16, 320 kb/s
---------------------------------------------------------------------------------
Stream mapping:
  Stream #0:0 -> #0:0 (h264 (native) -> h264 (libx264))
  Stream #0:1 -> #0:1 (ac3 (native) -> aac (libvo_aacenc))

======================================================================================
                               +--------------------+
                               |                    |
                Source [1] --> |       FFmpeg       | --> Destination [1]
                               |                    |
                               +--------------------+
======================================================================================

7:47:52 - Free space on destination disk at start 54,442 Mb.
7:47:52 - FFmpeg user script VIDEO option(s) included.

FFmpeg messages/warnings (limited):

[libx264 @ 03047000] using SAR=1/1
[libx264 @ 03047000] using cpu capabilities: MMX2 SSE SlowCTZ
[libx264 @ 03047000] profile High, level 4.1

video:21200kB audio:98kB subtitle:0kB other streams:0kB global headers:0kB
  ... muxing overhead: 0.053080%

7:52:00 - bitrate: min 0, max 13936, avg 9808 kbit/s
7:52:00 - Q-scale: min 0.0, max 23.0, avg 20.4

7:52:00 - Process successfully finished after 00:04:08 hms.

I don't get any buffer underflows or other warnings. As you can see, avarage qscale is 20.4 which is quite good.

If your source clip is a lead-in created with Ulead 3D, it's probably better to save a uncompressed avi from Ulead and from there convert to x264 with a silent audio stream. You could save a template for that to use with any output from Ulead 3D.

[vijm1972]

Hi Chris,

I wanted to create mp4 as per below information.

Commercial name : AVC
Format profile : Main@L4.0
Format settings : CABAC / 4 Ref Frames
Format settings, CABAC : Yes
Format settings, CABAC : Yes
Format settings, ReFrames : 4
Format settings, ReFrames : 4 frames
Format settings, GOP : M=1, N=15
Internet media type : video/H264


But after conversion I am getting file as per below information.

Commercial name : AVC
Format profile : Main@L4.0
Format settings : CABAC / 4 Ref Frames
Format settings, CABAC : Yes
Format settings, CABAC : Yes
Format settings, ReFrames : 4
Format settings, ReFrames : 4 frames
Format settings, GOP : M=4, N=15
Internet media type : video/H264


Only difference I have found in GOP settings M=4, is it possible to make it M=1?

I am using below FFMPEG Script.

-coder 1
-flags +loop
-cmp chroma
-partitions +parti8x8+parti4x4+partp8x8+partb8x8
-me_method umh
-subq 6
-me_range 16
-g 15
-keyint_min 1
-sc_threshold 40
-i_qfactor 0.71
-b_strategy 2
-qcomp 0.6
-qmin 10
-qmax 51
-qdiff 4
-direct-pred 1
-fast-pskip 1
-vprofile main -vlevel 4.0
-movflags faststart



Need your suggestion.

Thanks
Vijay

[Chris K]

Only difference I have found in GOP settings M=4, is it possible to make it M=1?

I don't know! I'm mainly familiar with the FFmpeg "shell" that encloses all the stuff and of course with the usage of Avanti. I only know the basics of encoding but I'm not an expert on the details of each individual codec.

You could try to add -bf 0 and see what that does. In my test with your script it says it encodes with 3 b-frames.

But since this is a general x264 question, you could also start a new thread on the subject and ask there.

Chris

[Chris K]

Current Avanti version 0.8.8 suffers under a recent FFmpeg change and a Zeranoe build issue
with builds released after January 12, 2015.

The color correction HSBC option will fail because of a deprecated mplayer filter which is now ported to libavfilter.

Avanti doesn't support FFmpeg builds older than January 2014 and checks out the FFmpeg build date for this.
Current Zeranoe builds do not include any build date so this check will fail.

You can find info on the last Zeranoe build that is usable with Avanti 0.8.8 and a link to the Zeranoe archive here ...

http://avanti.arrozcru.org/#download2

Hope to fix and release a new Avanti version soon.

Chris

[vijm1972]

Hi Chris,

I have source which is in packaged format and have 5 streams as per below format created from Omneon server.



.mov file is link between all the streams, when I select the .mov source file in source I can see all the files available on that folder. I wanted to transcode them into XDCAM HD MXF format. But when I start transcoding, Avanti gives error "unable to parse option value -1 as pixel format". Is there something I need to add into FFMPEG script?


Avanti able to import the file.




My output settings are as per below.




I am getting below error when I start transcoding.

[mov,mp4,m4a,3gp,3g2,mj2 @ 043cc060] stream 0, error opening alias:
... path='/srv/omneon/MediaGrid/pharos/FCP-HD-OUT/clip.dir/media.dir/XDCAM50_
... 013122N1_1.m2v', dir='media.dir', filename='XDCAM50_013122N1_1.m2v',
... volume='NONAME', nlvl_from=1, nlvl_to=2
[mov,mp4,m4a,3gp,3g2,mj2 @ 043cc060] stream 1, error opening alias:
... path='/srv/omneon/MediaGrid/pharos/FCP-HD-OUT/clip.dir/media.dir/XDCAM50_
... 013122N1_2.wav', dir='media.dir', filename='XDCAM50_013122N1_2.wav',
... volume='NONAME', nlvl_from=1, nlvl_to=2
[mov,mp4,m4a,3gp,3g2,mj2 @ 043cc060] stream 2, error opening alias:
... path='/srv/omneon/MediaGrid/pharos/FCP-HD-OUT/clip.dir/media.dir/XDCAM50_
... 013122N1_3.wav', dir='media.dir', filename='XDCAM50_013122N1_3.wav',
... volume='NONAME', nlvl_from=1, nlvl_to=2
[mov,mp4,m4a,3gp,3g2,mj2 @ 043cc060] stream 3, error opening alias:
... path='/srv/omneon/MediaGrid/pharos/FCP-HD-OUT/clip.dir/media.dir/XDCAM50_
... 013122N1_4.wav', dir='media.dir', filename='XDCAM50_013122N1_4.wav',
... volume='NONAME', nlvl_from=1, nlvl_to=2
[mov,mp4,m4a,3gp,3g2,mj2 @ 043cc060] stream 4, error opening alias:
... path='/srv/omneon/MediaGrid/pharos/FCP-HD-OUT/clip.dir/media.dir/XDCAM50_
... 013122N1_5.wav', dir='media.dir', filename='XDCAM50_013122N1_5.wav',
... volume='NONAME', nlvl_from=1, nlvl_to=2
[mov,mp4,m4a,3gp,3g2,mj2 @ 043cc060] Could not find codec parameters for stream 0 (Video: mpeg2video (xd5c / 0x63356478),
... none(tv), 1920x1080, 50034 kb/s): unspecified pixel format

Consider increasing the value for the 'analyzeduration' and 'probesize' options

Guessed Channel Layout for Input Stream #0.1 : stereo

Guessed Channel Layout for Input Stream #0.2 : stereo

Guessed Channel Layout for Input Stream #0.3 : stereo

Guessed Channel Layout for Input Stream #0.4 : stereo

Metadata_Read:
... creation_time : 2011-12-23 15:41:59
... encoder : Omneon libommedia 5,5,0,0 09-16-2009 13:38:34,ex={0,-1},rng={0,-1,1},exPre
... encoder-eng : Omneon libommedia 5,5,0,0 09-16-2009 13:38:34,ex={0,-1},rng={0,-1,1},exPre

Metadata_Read:
... creation_time : 2011-12-23 23:41:59
... handler_name : Apple Alias Data Handler
... encoder : MPEG
... timecode : 10:00:00:00

Metadata_Read:
... creation_time : 2011-12-23 23:41:59
... handler_name : Apple Alias Data Handler

Metadata_Read:
... creation_time : 2011-12-23 23:41:59
... handler_name : Apple Alias Data Handler

Metadata_Read:
... creation_time : 2011-12-23 23:41:59
... handler_name : Apple Alias Data Handler

Metadata_Read:
... creation_time : 2011-12-23 23:41:59
... handler_name : Apple Alias Data Handler

Metadata_Read:
... creation_time : 2011-12-23 23:41:59
... handler_name : Apple Alias Data Handler
... timecode : 10:00:00:00

[buffer @ 043ce5e0] Unable to parse option value "-1" as pixel format

Error opening filters!

18:47:47 - Process canceled by FFmpeg with 8 messages/warnings.


Thanks
Vijay

[Chris K]

Hi Chris,

I have source which is in packaged format and have 5 streams as per below format created from Omneon server.

.mov file is link between all the streams, when I select the .mov source file in source I can see all the files available on that folder. I wanted to transcode them into XDCAM HD MXF format. But when I start transcoding, Avanti gives error "unable to parse option value -1 as pixel format". Is there something I need to add into FFMPEG script?

I'm already surprised that FFmpeg can read the mov. It's a small reference file, I guess.

You can try to set a input pixel format at the "Colorspc." dropdown box. Probably yuv420p or yuv422p.

Chris

[vijm1972]

Hi Chris,

Thanks for the quick response and new upgrade 0.8.8

I had tried changing the colorspc settings, still getting same error. My ffmpeg build is also latest. (ffmpeg-20150131-git-f5722ba-win32-static)

Thanks
Vijay

[Chris K]

I had tried changing the colorspc settings, still getting same error. My ffmpeg build is also latest. (ffmpeg-20150131-git-f5722ba-win32-static)

Yes, my suggestion was wrong! It sets a output pixel format instead of forcing a input pixel format.

Try to add this at the [USER_VIDEO_OPTIONS] section.

-vfin "format=pix_fmts=yuv420p"

or -vfin "format=pix_fmts=yuv422p"

If that doesn't work, I'm out of ideas and can't help you further with this issue.

[vijm1972]

Not worked!!!

[Chris K]

Try to load the m2v file at "Source 1" and the first PCM audio file at "Source 2" and still use the -vfin "format=pix_fmts=yuv420p" command because the m2v file doesn't specify a pixel format.

[mov,mp4,m4a,3gp,3g2,mj2 @ 043cc060] Could not find codec parameters for stream 0
(Video: mpeg2video (xd5c / 0x63356478), none(tv), 1920x1080, 50034 kb/s): unspecified pixel format

[vijm1972]

Chris I am able to transcode the file. Is it possible to use all the 4 track while transcoding?

[Chris K]

Chris I am able to transcode the file. Is it possible to use all the 4 track while transcoding?

Yes you can!

1. Load the other three PCM files at the [USER_SOURCE_PATHS] section of the script editor.

Use the "Insert Source" button at the script editor to do that (select "as... user source path").

2. Double-click on the "Source 1" field to call up the source properties.

All files will then show up there. Then use the "Input stream mapping" option (<Alt> double-click on the streams) to select all the streams in the order from top to bottom.

You can first expand the source properties window (arrow button at the right of the header) to have a better view on all streams.

EDIT:

I wonder if it will work when you remux the video stream by setting video codec to "Copy Video"? That is, if the stream already fulfills your requirements.

[vijm1972]

Chris I am able to transcode the file with all the audio tracks. Needed your help on below.

In the same source .wav files we have stereo channels, means I have 4 .wav files with stereo channels as per below.

Every stream has

Stream 1 FL - RL
Stream 2 FC - LE
Stream 3 BL - BR
Stream 4 LT - RT

Now I wanted to transcode them to 8 streams. I am using below command but I am not getting proper audio output.

[Chris K]

Now I wanted to transcode them to 8 streams. I am using below command but I am not getting proper audio output.

You cannot do that all in one go. You'll have to first split the stereo streams to mono and then use the procedure you used for the 4 stereo streams to convert (or remux) the video with the 8 audio streams.

You can split the stereo streams with the following script. Since FFmpeg sees it as stereo streams, it doesn't care about FLE or BL. All streams are decoded as FL and FR. You have to gives them the proper channel name in the output file yourself.

[USER_VIDEO_OPTIONS]

-new

-y

-i "f:/source_audio_clip_1.wav"
-vn -das
-filter_complex "channelsplit=channel_layout=stereo[FL][FR]"
-map [FL] -o "f:/front_left.wav"
-map [FR] -o "f:/front_right.wav"

-i "f:/source_audio_clip_2.wav"
-vn -das
-filter_complex "channelsplit=channel_layout=stereo[FL][FR]"
-map [FL] -o "f:/front_center.wav"
-map [FR] -o "f:/low_frequency.wav"

Add the other two yourself with the proper source paths and output file names.

At the main page, disable video and set the audio settings as desired (probably PCM, mono). Also set the "Source 1" and "Destination" fields to NULL. This is required because all source and destination paths are given in the script.

This are the channel names as per FFmpeg ...

NAME DESCRIPTION
FL front left
FR front right
FC front center
LFE low frequency
BL back left
BR back right
FLC front left-of-center
FRC front right-of-center
BC back center
SL side left
SR side right
TC top center
TFL top front left
TFC top front center
TFR top front right
TBL top back left
TBC top back center
TBR top back right
DL downmix left
DR downmix right
WL wide left
WR wide right
SDL surround direct left
SDR surround direct right
LFE2 low frequency 2

[GoldenMeanie]

A couple more questions:

I wanted to ask, what does ReWrite PTS do?

Additionally, how does Avanti/FFMPEG/AVISynth handle "Keyframes" when rendering?

I'm having an issue with keyframes in another application (not because of Avanti) so I wondered how Avanti/FFMPEG handled them, if at all so I can resolve this issue.

[Chris K]

I wanted to ask, what does ReWrite PTS do?

It actually inserts the FFmpeg -fflags genpts command before the input. Afaik it overwrites the source time stamps starting at zero. It often helps when you try to remux something into a mkv container and get the following error ...

[matroska @ 02fa9020] Can't write packet with unknown timestamp

Additionally, how does Avanti/FFMPEG/AVISynth handle "Keyframes" when rendering?

x264 and mpeg2video for example let you set a gop size and b-frames. Each gop starts with a keyframe. It differs per codec and template created for it. Some use long gops, others use only keyframes.

Avanti doesn't do any magic with FFmpeg. It just supplies a command line and starts the process. I only know the basics of encoding. For details that are not directly related to the use Avanti you'd better start a thread and ask the gurus.

Chris

[GoldenMeanie]

I wanted to ask, what does ReWrite PTS do?

It actually inserts the FFmpeg -fflags genpts command before the input. Afaik it overwrites the source time stamps starting at zero. It often helps when you try to remux something into a mkv container and get the following error ...

[matroska @ 02fa9020] Can't write packet with unknown timestamp

Additionally, how does Avanti/FFMPEG/AVISynth handle "Keyframes" when rendering?

x264 and mpeg2video for example let you set a gop size and b-frames. Each gop starts with a keyframe. It differs per codec and template created for it. Some use long gops, others use only keyframes.

Avanti doesn't do any magic with FFmpeg. It just supplies a command line and starts the process. I only know the basics of encoding. For details that are not directly related to the use Avanti you'd better start a thread and ask the gurus.

Chris

Lol! Okay I guess I should have asked, what does PTS mean?

Thanks for your help. I obviously have more homework to do.

[GoldenMeanie]

Getting an odd error on my laptop for a simple function that shouldn't cause any issues.

I'm trying to extract the audio from my video clip but I keep getting the error:
No FFmpeg audio codec found for *. destination format

I've even deleted the .ini file and I get the same error.

My build for FFMPEG appears to be fine and initializes perfectly. I'm using one from May 6th, 2014.
This is the first I've ever seen of this. Usually I can rip the audio without a hitch and have done so a zillion times.
My desktop works just fine. It's using an older build from April.

Using Avanti 0.7.8. My configuration for this machine is Windows XP HOME SP4, Mobil Pentium 4 2.66ghz - 512mb ram.

[Chris K]

Getting an odd error on my laptop for a simple function that shouldn't cause any issues.

I'm trying to extract the audio from my video clip but I keep getting the error:
No FFmpeg audio codec found for *. destination format

It looks like your source clip contains a audio format for which no default file extension is stored in the codecs database. I can only help if you post a MediaInfo output of the clip that generates that error.

My build for FFMPEG appears to be fine and initializes perfectly. I'm using one from May 6th, 2014.
This is the first I've ever seen of this. Usually I can rip the audio without a hitch and have done so a zillion times.
My desktop works just fine. It's using an older build from April.

Using Avanti 0.7.8. My configuration for this machine is Windows XP HOME SP4, Mobil Pentium 4 2.66ghz - 512mb ram.

Although this doesn't has anything to do with the error, some of your system specs (memory) is far below minimum requirements. Not for Avanti but definitely for FFmpeg. You also use a rather old Avanti version (latest is 0.8.8).

On my old Athlon test machine I have 1.25gb which causes already delays and a occasional crash. On my Phenom I have 4gb. In the Avanti help at minimum system requirements it says...

Operating system: XP, Vista, Win 7, Win 8 (see note).
Processor: Pentium 4 or AMD Athlon running at 2.0 Ghz or higher.
Memory: minimum of 2Gb for XP, 4Gb for others.

EDIT:

Lol! Okay I guess I should have asked, what does PTS mean?

The full name of PTS is "presentation timestamp". It's what actually makes it possible to seek to any position in a media player.

Chris

[GoldenMeanie]

Getting an odd error on my laptop for a simple function that shouldn't cause any issues.

I'm trying to extract the audio from my video clip but I keep getting the error:
No FFmpeg audio codec found for *. destination format

It looks like your source clip contains a audio format for which no default file extension is stored in the codecs database. I can only help if you post a MediaInfo output of the clip that generates that error.

My build for FFMPEG appears to be fine and initializes perfectly. I'm using one from May 6th, 2014.
This is the first I've ever seen of this. Usually I can rip the audio without a hitch and have done so a zillion times.
My desktop works just fine. It's using an older build from April.

Using Avanti 0.7.8. My configuration for this machine is Windows XP HOME SP4, Mobil Pentium 4 2.66ghz - 512mb ram.

Although this doesn't has anything to do with the error, some of your system specs (memory) is far below minimum requirements. Not for Avanti but definitely for FFmpeg. You also use a rather old Avanti version (latest is 0.8.8).

On my old Athlon test machine I have 1.25gb which causes already delays and a occasional crash. On my Phenom I have 4gb. In the Avanti help at minimum system requirements it says...

Operating system: XP, Vista, Win 7, Win 8 (see note).
Processor: Pentium 4 or AMD Athlon running at 2.0 Ghz or higher.
Memory: minimum of 2Gb for XP, 4Gb for others.

EDIT:

Lol! Okay I guess I should have asked, what does PTS mean?

The full name of PTS is "presentation timestamp". It's what actually makes it possible to seek to any position in a media player.

Chris

Okay this is interesting because I used Avanti/FFMPEG to compress the audio on the source file. It's AAC/48000HZ/Stereo 320kbps. I will check the codecs database.

Also thanks for the memory spec issue. I will be buying 2GB of RAM for my machine soon, although FFMPEG and Avanti usually work very well on my system.

[troyvh]

I am trying to install this program and I have put the latest ffmpeg in the correct folder. I keep getting "unsupported ffmpeg version (older than January 2014)" error and program does not want to start up???? I am using window 7 computer.

edit:
Ok I downloaded a previous ffmpeg and it worked.