Re: [FFmpeg-user] Warning: VBV buffer size not set
On Thu, 13 Dec 2018, Carl Eugen Hoyos wrote: 2018-12-13 0:57 GMT+01:00, Hans Carlson : So, I'll rephrase Ulf's original question... how do you avoid this warning while doing a stream COPY (remux)? You cannot, as the maximum bit-rate cannot be set when remuxing. The muxer has to assume that your input confirms to the relevant specification. Note that command lines without "-f" make it impossible for ffmpeg to even understand which standard you want. So why did re-muxing work fine without any warnings before this change? https://github.com/FFmpeg/FFmpeg/commit/079b5d4ef888bd42bf0147a6d964b8bc9ec0f3c5#diff-00823e6d5f4d3807869c905426e6bdd1 Prior to this change, the buffer size (along with other values) were copied from the decoding context and used for the encoding context. This was done in ffmpeg.c:transcode_init() in an if statement specifically for stream_copy: 2903 enc_ctx->bit_rate = dec_ctx->bit_rate; 2904 enc_ctx->rc_max_rate= dec_ctx->rc_max_rate; 2905 enc_ctx->rc_buffer_size = dec_ctx->rc_buffer_size; 2906 enc_ctx->field_order= dec_ctx->field_order; NOTE: Line numbers are based on rev: 079b5d4ef888bd42bf0147a6d964b8bc9ec0f3c5 Why is it now "impossible" to get that same information from the decoding context? Forgive my ignorance, but from my perspective, it USED to work and now it doesn't work. ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
Re: [FFmpeg-user] Warning: VBV buffer size not set
2018-12-13 0:57 GMT+01:00, Hans Carlson : > On Wed, 12 Dec 2018, Carl Eugen Hoyos wrote: > >> 2018-12-08 22:28 GMT+01:00, Ulf Zibis : >> >>> with: >>> ffmpeg -i "concat:CYD-001.vob|CYD-002.vob|..." -c copy CYD_copy.vob >>> I get the warning: >>> [svcd @ 0x56377151e140] VBV buffer size not set, using default size of >>> 130KB >>> If you want the mpeg file to be compliant to some specification >>> Like DVD, VCD or others, make sure you set the correct buffer size >>> >>> How can I aviod the warning? >> >> Use the target option. > > If the target option is used WITHOUT "-c copy", then it's true you won't > see this warning. But then it's doing a re-encode... NOT a copy. The > problem comes when trying to do a COPY and the target option doesn't > help in that case. > > So, I'll rephrase Ulf's original question... how do you avoid this warning > while doing a stream COPY (remux)? You cannot, as the maximum bit-rate cannot be set when remuxing. The muxer has to assume that your input confirms to the relevant specification. Note that command lines without "-f" make it impossible for ffmpeg to even understand which standard you want. Carl Eugen ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
Re: [FFmpeg-user] Warning: VBV buffer size not set
On Wed, 12 Dec 2018, Carl Eugen Hoyos wrote: 2018-12-08 22:28 GMT+01:00, Ulf Zibis : with: ffmpeg -i "concat:CYD-001.vob|CYD-002.vob|..." -c copy CYD_copy.vob I get the warning: [svcd @ 0x56377151e140] VBV buffer size not set, using default size of 130KB If you want the mpeg file to be compliant to some specification Like DVD, VCD or others, make sure you set the correct buffer size How can I aviod the warning? Use the target option. If the target option is used WITHOUT "-c copy", then it's true you won't see this warning. But then it's doing a re-encode... NOT a copy. The problem comes when trying to do a COPY and the target option doesn't help in that case. So, I'll rephrase Ulf's original question... how do you avoid this warning while doing a stream COPY (remux)? FYI. You get the same result here with or without "-target XXX". $ ffmpeg -i TEST.mpg -target ntsc-dvd -codec copy -f vob - > /dev/null ffmpeg version N-92681-g0e833f615b Copyright (c) 2000-2018 the FFmpeg developers built with gcc 7 (GCC) configuration: --disable-optimizations --disable-stripping --enable-static --disable-shared --disable-ffplay libavutil 56. 24.101 / 56. 24.101 libavcodec 58. 42.100 / 58. 42.100 libavformat58. 24.100 / 58. 24.100 libavdevice58. 6.101 / 58. 6.101 libavfilter 7. 46.101 / 7. 46.101 libswscale 5. 4.100 / 5. 4.100 libswresample 3. 4.100 / 3. 4.100 Input #0, mpeg, from 'TEST.mpg': Duration: 00:00:09.88, start: 0.533367, bitrate: 142 kb/s Stream #0:0[0x1e0]: Video: mpeg2video (Main), yuv420p(tv, progressive), 720x480 [SAR 32:27 DAR 16:9], 29.97 fps, 29.97 tbr, 90k tbn, 59.94 tbc [vob @ 0x4000e00] VBV buffer size not set, using default size of 230KB If you want the mpeg file to be compliant to some specification Like DVD, VCD or others, make sure you set the correct buffer size Output #0, vob, to 'pipe:': Metadata: encoder : Lavf58.24.100 Stream #0:0: Video: mpeg2video (Main), yuv420p(tv, progressive), 720x480 [SAR 32:27 DAR 16:9], q=2-31, 6000 kb/s, 29.97 fps, 29.97 tbr, 90k tbn, 29.97 tbc Stream mapping: Stream #0:0 -> #0:0 (copy) Press [q] to stop, [?] for help [vob @ 0x4000e00] Timestamps are unset in a packet for stream 0. This is deprecated and will stop working in the future. Fix your code to set the timestamps properly frame= 300 fps=0.0 q=-1.0 Lsize= 172kB time=00:00:09.94 bitrate= 141.7kbits/s speed=4.21e+03x video:169kB audio:0kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 1.759272% ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
[FFmpeg-user] consult volume
hello, how can you check the volume number of a file?, to change it useful -filter "volume", but I do not know how to consult the original volume thanks ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
Re: [FFmpeg-user] Trimming email quotes
Am 12.12.18 um 19:46 schrieb Carl Zwanzig: > And also cut the quoted footers since each list email gets a fresh one people all over the world are not capable to basically handle mail clients, sad but true ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
[FFmpeg-user] Issue on flickering on packaging side
Hi , We are new for ffmpeg and did some live streaming for IPTV . Now we are moving to production while packaging side getting problem on it . Transcoding working well. Using multicast to get stream input to one server transcoded then send to packager(another server ) .Problem was getting flickering when streaming in any devices and getting million of continuity counter error .we tested in networking side also whether packet loss or not that one also fine .we are struck on this stage .any one please suggest me ? Transcoding : ffmpeg -i udp://x.x.x.x.:7116?fifo_size=2500_nonfatal=0_size=5000=10.55.1.80 -f mpegts udp://y.y.y.y:6808?localaddr=192.168.29.52_size=1316 Packaging : ffmpeg -i udp://y.y.y.y:6808?fifo_size=2500_nonfatal=1_size=5000=192.168.29.102 -map 0:p:1:3 -map 0:p:1:4 -c copy /var/www/html/Live/test.mp4 Sample screenshot Flickering Thanks & Regards Vasanth Sent from my iPhone ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
Re: [FFmpeg-user] Warning: VBV buffer size not set
2018-12-12 20:11 GMT+01:00, Carl Eugen Hoyos : > 2018-12-08 22:28 GMT+01:00, Ulf Zibis : > >> with: >> ffmpeg -i "concat:CYD-001.vob|CYD-002.vob|..." -c copy CYD_copy.vob (My answer wasn't wrong but your command line does not make much sense.) If your command succeeds (ie you ignore Hans' warning but still get a useful output file) the following will also work and has many advantages (apart from not showing a muxer warning): cat CYD-001.vob CYD-002.vob ... >CYD_copy.vob Carl Eugen ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
Re: [FFmpeg-user] Warning: VBV buffer size not set
2018-12-08 22:28 GMT+01:00, Ulf Zibis : > with: > ffmpeg -i "concat:CYD-001.vob|CYD-002.vob|..." -c copy CYD_copy.vob > I get the warning: > [svcd @ 0x56377151e140] VBV buffer size not set, using default size of 130KB > If you want the mpeg file to be compliant to some specification > Like DVD, VCD or others, make sure you set the correct buffer size > > How can I aviod the warning? Use the target option. Carl Eugen ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
Re: [FFmpeg-user] Warning: VBV buffer size not set
On Sat, 8 Dec 2018, Ulf Zibis wrote:
with:
ffmpeg -i "concat:CYD-001.vob|CYD-002.vob|..." -c copy CYD_copy.vob
I get the warning:
[svcd @ 0x56377151e140] VBV buffer size not set, using default size of 130KB
If you want the mpeg file to be compliant to some specification
Like DVD, VCD or others, make sure you set the correct buffer size
How can I aviod the warning?
Two things...
1) This has nothing to do with the warning you're receiving, but you
should NOT concat VOB's like that. It works... "sometimes"... which is why
many people unfortunately suggest it, but it quite often does not work.
VOB's are not simple MPEG files. You'd be better off using one of the
following methods:
- tools/dvd2concat (script included with ffmpeg source)
- mpv ... --stream-dump ...
- mplayer ... --dumpstream ...
You'll need to read the man pages for each to get the exact command and
options.
2) With regards to the VBV buffer size warning... Unfortunately, I don't
have a good answer, only a little history. This has been an issue for a
few years and I wish I knew how to fix it. I've resorted to patching the
source to use a higher default buffer size but I know that's not the
correct or even a good solution. I've always intended to try and dig
deeper to find the cause, but lack of time and experience with the code
has always halted my progress.
The "VBV buffer size not set" warning itself is nothing new... it's been
in the code for several years as far as I can tell. But I never noticed
the warning until around version ~3.0... specifically, this change:
https://github.com/FFmpeg/FFmpeg/commit/079b5d4ef888bd42bf0147a6d964b8bc9ec0f3c5#diff-00823e6d5f4d3807869c905426e6bdd1
After that change, the VBV warning started to get displayed when doing a
stream copy on mpeg files. From what I can tell, in versions prior to
this change ffmpeg determined the "VBV buffer size" for the destination
file by looking at the source file. But when the above change happened
that value was no longer available, so internally no buffer size was found
for the destination file, which triggers the "VBV buffer size not set"
warning message.
Starting with the above change (or there abouts), ffmpeg started to use a
different internal structure ("side data") to store some of the
information about a file. My guess as to the problem with the VBV buffer
warning is that it has something to do with how this "side data" is
populated, but I've never dug deep enough to figure that out.
___
ffmpeg-user mailing list
ffmpeg-user@ffmpeg.org
http://ffmpeg.org/mailman/listinfo/ffmpeg-user
To unsubscribe, visit link above, or email
ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
Re: [FFmpeg-user] Trimming email quotes
On 12/12/18, Carl Zwanzig wrote: > Any chance folks could start editing their replies to remove the extraneous > quoting? > No, we like to waste resources. ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
[FFmpeg-user] Trimming email quotes
Any chance folks could start editing their replies to remove the extraneous quoting? On 12/12/2018 10:39 AM, Ronak wrote: On Dec 12, 2018, at 12:10 PM, Paul B Mahol wrote: On 12/12/18, Ronak wrote: On Dec 12, 2018, at 11:36 AM, Paul B Mahol wrote: On 12/12/18, Ronak wrote: On Dec 12, 2018, at 11:26 AM, Paul B Mahol wrote: On 12/12/18, Ronak wrote: On Dec 12, 2018, at 8:32 AM, Nicolas George wrote: Ronak (2018-12-11): [55 lines of quotes] [one-line response] And also cut the quoted footers since each list email gets a fresh one. Thanks, z! ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
Re: [FFmpeg-user] Dynaudnorm & earwax filters
On 12/12/18, Ronak wrote: > > >> On Dec 12, 2018, at 12:10 PM, Paul B Mahol wrote: >> >> On 12/12/18, Ronak wrote: >>> >>> On Dec 12, 2018, at 11:36 AM, Paul B Mahol wrote: On 12/12/18, Ronak wrote: > >> On Dec 12, 2018, at 11:26 AM, Paul B Mahol wrote: >> >> On 12/12/18, Ronak wrote: >>> >>> On Dec 12, 2018, at 8:32 AM, Nicolas George wrote: Ronak (2018-12-11): > Ok thanks. I tried to use this filter in my iOS code; but I'm > getting > errors with an error code -35. > > This is my code that tries to write data into the filter graph and > reads it back; what am I doing wrong? I do not read whatever language that is, but at the very least your code is missing the translation error code -> error message. >>> >>> I found out what my problem is; it's that the dynaudnorm filter is >>> returning >>> EAGAIN; which means I need to send it more PCM frames. >>> >>> Now, I'm trying to integrate this filter into a real time player >>> context; >>> and I would like to avoid audio artifacts. I've been playing with >>> various >>> options that the filter has; but I can't seem to find one where it >>> would >>> work better in the real time context. >>> >>> Does anyone know what the correct parameters would be so it works >>> frame >>> by >>> frame or in a much smaller frame size so we can avoid audio >>> artifacts? >>> Alternatively, is there another ffmpeg filter better suited to real >>> time >>> dynamic range compression or volume normalization? >>> >> >> If you read documentation of filter options you would know. > > I already did and tried all sorts of things. I've tried options like: > "f=500:g=3:m=10:n=1:b=1", "f=40:g=3:m=10:n=1:b=1". I've even tried the > extreme: "f=8000:g=3:m=10:n=1:b=1" > > But I still get back lots of EAGAIN. That's normal, if you insist on 0 latency look at something else. Other players like mpv, handle it fine. >>> >>> Ok. One last thing is it seems like the filter is spitting out lots of >>> pops >>> and crackles when I can get it to return audio frames back out. >>> >>> Do you know why that would be? I changed all my arguments to just be >>> f="1000" since I thought my options would be causing this. But it's not. >>> >>> Just in case it helps, I am sending in FLTP which is being resampled by >>> the >>> rwresample filter to S32. I don't think that would be a factor in this >>> right? >>> >> >> You should send only DBL to this filter. > > Sorry I misquoted. > > [volume normalization @ 0x7fa4c860dd80] auto-inserting filter > 'auto_resampler_0' between the filter 'input' and the filter 'volume > normalization' > [auto_resampler_0 @ 0x7fa4c8610940] ch:2 chl:stereo fmt:fltp r:44100Hz -> > ch:2 chl:stereo fmt:dblp r:44100Hz > > It is being resampled to DBLP. > > Besides doing a whole bunch of trial and error, are there any recommended > options to use here? > > I'm writing one frame of PCM audio into the filter at a time, within my > playback audio graph. > I can not guess, need to look at source code. ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
Re: [FFmpeg-user] Dynaudnorm & earwax filters
> On Dec 12, 2018, at 12:10 PM, Paul B Mahol wrote: > > On 12/12/18, Ronak wrote: >> >> >>> On Dec 12, 2018, at 11:36 AM, Paul B Mahol wrote: >>> >>> On 12/12/18, Ronak wrote: > On Dec 12, 2018, at 11:26 AM, Paul B Mahol wrote: > > On 12/12/18, Ronak wrote: >> >> >>> On Dec 12, 2018, at 8:32 AM, Nicolas George wrote: >>> >>> Ronak (2018-12-11): Ok thanks. I tried to use this filter in my iOS code; but I'm getting errors with an error code -35. This is my code that tries to write data into the filter graph and reads it back; what am I doing wrong? >>> >>> I do not read whatever language that is, but at the very least your >>> code >>> is missing the translation error code -> error message. >>> >> >> I found out what my problem is; it's that the dynaudnorm filter is >> returning >> EAGAIN; which means I need to send it more PCM frames. >> >> Now, I'm trying to integrate this filter into a real time player >> context; >> and I would like to avoid audio artifacts. I've been playing with >> various >> options that the filter has; but I can't seem to find one where it >> would >> work better in the real time context. >> >> Does anyone know what the correct parameters would be so it works frame >> by >> frame or in a much smaller frame size so we can avoid audio artifacts? >> Alternatively, is there another ffmpeg filter better suited to real >> time >> dynamic range compression or volume normalization? >> > > If you read documentation of filter options you would know. I already did and tried all sorts of things. I've tried options like: "f=500:g=3:m=10:n=1:b=1", "f=40:g=3:m=10:n=1:b=1". I've even tried the extreme: "f=8000:g=3:m=10:n=1:b=1" But I still get back lots of EAGAIN. >>> >>> That's normal, if you insist on 0 latency look at something else. >>> Other players like mpv, handle it fine. >> >> Ok. One last thing is it seems like the filter is spitting out lots of pops >> and crackles when I can get it to return audio frames back out. >> >> Do you know why that would be? I changed all my arguments to just be >> f="1000" since I thought my options would be causing this. But it's not. >> >> Just in case it helps, I am sending in FLTP which is being resampled by the >> rwresample filter to S32. I don't think that would be a factor in this >> right? >> > > You should send only DBL to this filter. Sorry I misquoted. [volume normalization @ 0x7fa4c860dd80] auto-inserting filter 'auto_resampler_0' between the filter 'input' and the filter 'volume normalization' [auto_resampler_0 @ 0x7fa4c8610940] ch:2 chl:stereo fmt:fltp r:44100Hz -> ch:2 chl:stereo fmt:dblp r:44100Hz It is being resampled to DBLP. Besides doing a whole bunch of trial and error, are there any recommended options to use here? I'm writing one frame of PCM audio into the filter at a time, within my playback audio graph. > ___ > ffmpeg-user mailing list > ffmpeg-user@ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > > To unsubscribe, visit link above, or email > ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe". ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
Re: [FFmpeg-user] running configure on SPARC
2018-12-12 16:25 GMT+01:00, Eric Thomas : > I am trying to build a set of shared libraries. I have successfully run > these different configure commands: > > bash ./configure --prefix=/myhome/ffmpeg/FFmpeg-4.0.3 Only current FFmpeg git head is supported here, sorry. Please find out what "top-posting" means and avoid it here, Carl Eugen ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
Re: [FFmpeg-user] Dynaudnorm & earwax filters
On 12/12/18, Ronak wrote: > > >> On Dec 12, 2018, at 11:36 AM, Paul B Mahol wrote: >> >> On 12/12/18, Ronak wrote: >>> On Dec 12, 2018, at 11:26 AM, Paul B Mahol wrote: On 12/12/18, Ronak wrote: > > >> On Dec 12, 2018, at 8:32 AM, Nicolas George wrote: >> >> Ronak (2018-12-11): >>> Ok thanks. I tried to use this filter in my iOS code; but I'm getting >>> errors with an error code -35. >>> >>> This is my code that tries to write data into the filter graph and >>> reads it back; what am I doing wrong? >> >> I do not read whatever language that is, but at the very least your >> code >> is missing the translation error code -> error message. >> > > I found out what my problem is; it's that the dynaudnorm filter is > returning > EAGAIN; which means I need to send it more PCM frames. > > Now, I'm trying to integrate this filter into a real time player > context; > and I would like to avoid audio artifacts. I've been playing with > various > options that the filter has; but I can't seem to find one where it > would > work better in the real time context. > > Does anyone know what the correct parameters would be so it works frame > by > frame or in a much smaller frame size so we can avoid audio artifacts? > Alternatively, is there another ffmpeg filter better suited to real > time > dynamic range compression or volume normalization? > If you read documentation of filter options you would know. >>> >>> I already did and tried all sorts of things. I've tried options like: >>> "f=500:g=3:m=10:n=1:b=1", "f=40:g=3:m=10:n=1:b=1". I've even tried the >>> extreme: "f=8000:g=3:m=10:n=1:b=1" >>> >>> But I still get back lots of EAGAIN. >> >> That's normal, if you insist on 0 latency look at something else. >> Other players like mpv, handle it fine. > > Ok. One last thing is it seems like the filter is spitting out lots of pops > and crackles when I can get it to return audio frames back out. > > Do you know why that would be? I changed all my arguments to just be > f="1000" since I thought my options would be causing this. But it's not. > > Just in case it helps, I am sending in FLTP which is being resampled by the > rwresample filter to S32. I don't think that would be a factor in this > right? > You should send only DBL to this filter. ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
Re: [FFmpeg-user] Dynaudnorm & earwax filters
> On Dec 12, 2018, at 11:36 AM, Paul B Mahol wrote: > > On 12/12/18, Ronak wrote: >> >>> On Dec 12, 2018, at 11:26 AM, Paul B Mahol wrote: >>> >>> On 12/12/18, Ronak wrote: > On Dec 12, 2018, at 8:32 AM, Nicolas George wrote: > > Ronak (2018-12-11): >> Ok thanks. I tried to use this filter in my iOS code; but I'm getting >> errors with an error code -35. >> >> This is my code that tries to write data into the filter graph and >> reads it back; what am I doing wrong? > > I do not read whatever language that is, but at the very least your code > is missing the translation error code -> error message. > I found out what my problem is; it's that the dynaudnorm filter is returning EAGAIN; which means I need to send it more PCM frames. Now, I'm trying to integrate this filter into a real time player context; and I would like to avoid audio artifacts. I've been playing with various options that the filter has; but I can't seem to find one where it would work better in the real time context. Does anyone know what the correct parameters would be so it works frame by frame or in a much smaller frame size so we can avoid audio artifacts? Alternatively, is there another ffmpeg filter better suited to real time dynamic range compression or volume normalization? >>> >>> If you read documentation of filter options you would know. >> >> I already did and tried all sorts of things. I've tried options like: >> "f=500:g=3:m=10:n=1:b=1", "f=40:g=3:m=10:n=1:b=1". I've even tried the >> extreme: "f=8000:g=3:m=10:n=1:b=1" >> >> But I still get back lots of EAGAIN. > > That's normal, if you insist on 0 latency look at something else. > Other players like mpv, handle it fine. Ok. One last thing is it seems like the filter is spitting out lots of pops and crackles when I can get it to return audio frames back out. Do you know why that would be? I changed all my arguments to just be f="1000" since I thought my options would be causing this. But it's not. Just in case it helps, I am sending in FLTP which is being resampled by the rwresample filter to S32. I don't think that would be a factor in this right? > ___ > ffmpeg-user mailing list > ffmpeg-user@ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > > To unsubscribe, visit link above, or email > ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe". ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
Re: [FFmpeg-user] Dynaudnorm & earwax filters
On 12/12/18, Ronak wrote: > >> On Dec 12, 2018, at 11:26 AM, Paul B Mahol wrote: >> >> On 12/12/18, Ronak wrote: >>> >>> On Dec 12, 2018, at 8:32 AM, Nicolas George wrote: Ronak (2018-12-11): > Ok thanks. I tried to use this filter in my iOS code; but I'm getting > errors with an error code -35. > > This is my code that tries to write data into the filter graph and > reads it back; what am I doing wrong? I do not read whatever language that is, but at the very least your code is missing the translation error code -> error message. >>> >>> I found out what my problem is; it's that the dynaudnorm filter is >>> returning >>> EAGAIN; which means I need to send it more PCM frames. >>> >>> Now, I'm trying to integrate this filter into a real time player context; >>> and I would like to avoid audio artifacts. I've been playing with various >>> options that the filter has; but I can't seem to find one where it would >>> work better in the real time context. >>> >>> Does anyone know what the correct parameters would be so it works frame >>> by >>> frame or in a much smaller frame size so we can avoid audio artifacts? >>> Alternatively, is there another ffmpeg filter better suited to real time >>> dynamic range compression or volume normalization? >>> >> >> If you read documentation of filter options you would know. > > I already did and tried all sorts of things. I've tried options like: > "f=500:g=3:m=10:n=1:b=1", "f=40:g=3:m=10:n=1:b=1". I've even tried the > extreme: "f=8000:g=3:m=10:n=1:b=1" > > But I still get back lots of EAGAIN. That's normal, if you insist on 0 latency look at something else. Other players like mpv, handle it fine. ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
Re: [FFmpeg-user] Dynaudnorm & earwax filters
> On Dec 12, 2018, at 11:26 AM, Paul B Mahol wrote: > > On 12/12/18, Ronak wrote: >> >> >>> On Dec 12, 2018, at 8:32 AM, Nicolas George wrote: >>> >>> Ronak (2018-12-11): Ok thanks. I tried to use this filter in my iOS code; but I'm getting errors with an error code -35. This is my code that tries to write data into the filter graph and reads it back; what am I doing wrong? >>> >>> I do not read whatever language that is, but at the very least your code >>> is missing the translation error code -> error message. >>> >> >> I found out what my problem is; it's that the dynaudnorm filter is returning >> EAGAIN; which means I need to send it more PCM frames. >> >> Now, I'm trying to integrate this filter into a real time player context; >> and I would like to avoid audio artifacts. I've been playing with various >> options that the filter has; but I can't seem to find one where it would >> work better in the real time context. >> >> Does anyone know what the correct parameters would be so it works frame by >> frame or in a much smaller frame size so we can avoid audio artifacts? >> Alternatively, is there another ffmpeg filter better suited to real time >> dynamic range compression or volume normalization? >> > > If you read documentation of filter options you would know. I already did and tried all sorts of things. I've tried options like: "f=500:g=3:m=10:n=1:b=1", "f=40:g=3:m=10:n=1:b=1". I've even tried the extreme: "f=8000:g=3:m=10:n=1:b=1" But I still get back lots of EAGAIN. > ___ > ffmpeg-user mailing list > ffmpeg-user@ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > > To unsubscribe, visit link above, or email > ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe". ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
Re: [FFmpeg-user] Dynaudnorm & earwax filters
On 12/12/18, Ronak wrote: > > >> On Dec 12, 2018, at 8:32 AM, Nicolas George wrote: >> >> Ronak (2018-12-11): >>> Ok thanks. I tried to use this filter in my iOS code; but I'm getting >>> errors with an error code -35. >>> >>> This is my code that tries to write data into the filter graph and >>> reads it back; what am I doing wrong? >> >> I do not read whatever language that is, but at the very least your code >> is missing the translation error code -> error message. >> > > I found out what my problem is; it's that the dynaudnorm filter is returning > EAGAIN; which means I need to send it more PCM frames. > > Now, I'm trying to integrate this filter into a real time player context; > and I would like to avoid audio artifacts. I've been playing with various > options that the filter has; but I can't seem to find one where it would > work better in the real time context. > > Does anyone know what the correct parameters would be so it works frame by > frame or in a much smaller frame size so we can avoid audio artifacts? > Alternatively, is there another ffmpeg filter better suited to real time > dynamic range compression or volume normalization? > If you read documentation of filter options you would know. ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
Re: [FFmpeg-user] Dynaudnorm & earwax filters
> On Dec 12, 2018, at 8:32 AM, Nicolas George wrote: > > Ronak (2018-12-11): >> Ok thanks. I tried to use this filter in my iOS code; but I'm getting >> errors with an error code -35. >> >> This is my code that tries to write data into the filter graph and >> reads it back; what am I doing wrong? > > I do not read whatever language that is, but at the very least your code > is missing the translation error code -> error message. > I found out what my problem is; it's that the dynaudnorm filter is returning EAGAIN; which means I need to send it more PCM frames. Now, I'm trying to integrate this filter into a real time player context; and I would like to avoid audio artifacts. I've been playing with various options that the filter has; but I can't seem to find one where it would work better in the real time context. Does anyone know what the correct parameters would be so it works frame by frame or in a much smaller frame size so we can avoid audio artifacts? Alternatively, is there another ffmpeg filter better suited to real time dynamic range compression or volume normalization? > Regards, > > -- > Nicolas George > ___ > ffmpeg-user mailing list > ffmpeg-user@ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > > To unsubscribe, visit link above, or email > ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe". ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
Re: [FFmpeg-user] Question about using Webm/Matroska container in audio transconding
2018-12-12 16:52 GMT+01:00, Ivan : >>The ogg format apparently allows for this feature, note that the >>format has so many issues that it cannot be seriously recommended. > > Just to be sure I understand you say ogg container is not good, right? Can > you bit elaborate what is the problem if I'm using it for audio only media? There may be no problem for you, most people who implemented support were not happy with it and the fact that it needs special (from development- side) for every single codec seems like a big disadvantage. > I have been using it (opus+ogg) for couple of years and have not met any > significant problems. "Never change a running system." ;-) Carl Eugen ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
Re: [FFmpeg-user] Question about using Webm/Matroska container in audio transconding
>The ogg format apparently allows for this feature, note that the >format has so many issues that it cannot be seriously recommended. > Just to be sure I understand you say ogg container is not good, right? Can you bit elaborate what is the problem if I'm using it for audio only media? I have been using it (opus+ogg) for couple of years and have not met any significant problems. Thanks Ivan ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
Re: [FFmpeg-user] running configure on SPARC
Hey Carl, I am trying to build a set of shared libraries. I have successfully run these different configure commands: bash ./configure --prefix=/myhome/ffmpeg/FFmpeg-4.0.3 --enable-shared bash ./configure --prefix=/myhome/ffmpeg/FFmpeg-4.0.3 --enable-shared --extra-cflags="-fPIC" bash ./configure --prefix=/myhome/ffmpeg/FFmpeg-4.0.3 --enable-shared --extra-cflags="-fPIC" --enable-pic In each case, the subsequent 'make' call returns with this error: [myhome/ffmpeg/FFmpeg-4.0.3]$ make Text relocation remains referenced against symbol offsetin file .data(section) 0x0 libavutil/../compat/atomics/pthread/stdatomic.o .data(section) 0xc libavutil/../compat/atomics/pthread/stdatomic.o .data(section) 0x14 libavutil/../compat/atomics/pthread/stdatomic.o .data(section) 0x20 libavutil/../compat/atomics/pthread/stdatomic.o pthread_mutex_lock 0x4 libavutil/../compat/atomics/pthread/stdatomic.o pthread_mutex_lock 0x8 libavutil/../compat/atomics/pthread/stdatomic.o pthread_mutex_lock 0x18 libavutil/../compat/atomics/pthread/stdatomic.o pthread_mutex_lock 0x1c libavutil/../compat/atomics/pthread/stdatomic.o ld: fatal: relocations remain against allocatable but non-writable sections collect2: ld returned 1 exit status make: *** [libavutil/libavutil.so.] Error 1 Can you point me in a right direction? I have not received any responses to my SuperUser post.. Thanks! On Wed, Dec 5, 2018 at 11:57 AM Carl Eugen Hoyos wrote: > 2018-12-05 17:54 GMT+01:00, Eric Thomas : > > > I had our sys-ad install gnu sed (v4.5), and I modified the 3 instances > of > > "sed -E" in the configure script. > > This is not necessary anymore, configure was fixed. > > > It now runs to completion - although it takes about 6 minutes to > complete. > > Which version of FFmpeg is this? > It took hours here before a recent fix, it is significantly faster now. > > Carl Eugen > ___ > ffmpeg-user mailing list > ffmpeg-user@ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > > To unsubscribe, visit link above, or email > ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe". ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
Re: [FFmpeg-user] Question about using Webm/Matroska container in audio transconding
2018-12-12 8:30 GMT+01:00, Ivan Zderadicka : > > On 11/12/18 18:41, Carl Eugen Hoyos wrote: >> 2018-12-11 18:26 GMT+01:00, Ivan : >> >>> just curious why it then works with ogg container and not in webm. The formats are different, you need to seek back to write the matroska duration, see line 2591 (and 2638 in the same block) in current libavformat/matroskaenc.c, this is not a limitation of FFmpeg. >> I am not convinced it works, would need console output. > > Please see attached listing of console session - console_output.txt - > file with ogg container does have duration, file with webm does not (as > can be seen in ffprobe listing), but both are created through pipe. The ogg format apparently allows for this feature, note that the format has so many issues that it cannot be seriously recommended. Carl Eugen ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
Re: [FFmpeg-user] Dynaudnorm & earwax filters
Ronak (2018-12-11): > Ok thanks. I tried to use this filter in my iOS code; but I'm getting > errors with an error code -35. > > This is my code that tries to write data into the filter graph and > reads it back; what am I doing wrong? I do not read whatever language that is, but at the very least your code is missing the translation error code -> error message. Regards, -- Nicolas George signature.asc Description: Digital signature ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
Re: [FFmpeg-user] Question about using Webm/Matroska container in audio transconding
On Wed, Dec 12, 2018 at 08:30:27 +0100, Ivan Zderadicka wrote: > ivan@ivan-ThinkPad-T460:~/tmp$ ffprobe test1.webm [...] > Input #0, matroska,webm, from 'test1.webm': > Metadata: > title : 10.kapitola > encoder : Lavf56.40.101 > Duration: N/A, start: 0.007000, bitrate: N/A > Stream #0:0: Audio: opus, 48000 Hz, stereo, fltp (default) I get: $ ffmpeg -i test_time.webm [...] Input #0, matroska,webm, from 'test_time.webm': Metadata: ENCODER : Lavf58.23.102 Duration: 00:00:05.00, start: -0.007000, bitrate: 27 kb/s Stream #0:0: Audio: opus, 48000 Hz, mono, fltp (default) Metadata: ENCODER : Lavc58.41.101 libopus My suggestion: > ffmpeg version 2.8.15-0ubuntu0.16.04.1 Copyright (c) 2000-2018 the FFmpeg > developers > built with gcc 5.4.0 (Ubuntu 5.4.0-6ubuntu1~16.04.10) 20160609 Your version of ffmpeg is ancient. Your issue may have long been solved. Please get hold of a recent version of ffmpeg. If you can't find a Ubunuto repository which provides latest builds, grab a binary from here: https://johnvansickle.com/ffmpeg/ (Preferred: Left column, git builds - not the releases.) I have no idea whether the file creation or the interpretation of the created file is different in recent ffmpeg (I assume the former), but it just works with webm. Moritz ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
