> On Oct 11, 2022, at 1:45 AM, Erik Dobberkau wrote:
>
>>> Is there a way to avoid having FFmpeg inject those additional silent
>> packets while concatenating?
>>
>> Here is my FFmpeg information: […]
>>
>
> Is your audio encoded using AAC? Your uncut console output would tell had
> it not be
> On Oct 10, 2022, at 10:43 PM, Ronak via ffmpeg-user
> wrote:
>
> Hi,
>
> I’m trying to concatenate a bunch of audio only MP4s together using FFmpeg’s
> concat demuxer.
>
> When I tested this with concaving 4 MP4 files into a larger one, FFmpeg seems
> to inj
those additional silent packets
while concatenating?
Does FFmpeg inject this silence regardless of what output there is? (Raw AAC vs
AAC in MP4).
Ronak
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> On Oct 3, 2019, at 5:27 PM, Carl Eugen Hoyos wrote:
>
> Am Do., 3. Okt. 2019 um 06:49 Uhr schrieb Ronak via ffmpeg-user
> :
>> I’m writing a C++ program to validate the integrity of a Fragmented MP4 file
>> containing AAC audio.
>
> Note that when using the l
> On Oct 3, 2019, at 12:48 AM, Ronak via ffmpeg-user
> wrote:
>
> Hi,
>
> I’m writing a C++ program to validate the integrity of a Fragmented MP4 file
> containing AAC audio.
> This program would parse the FMP4 file, read each audio packet, attachment
> ADTS head
MP4 atoms correctly, however
I’m having issues figuring out the best way to attach the ADTS headers.
The best sample I was able to find was:
https://patchwork.ffmpeg.org/patch/3184/
<https://patchwork.ffmpeg.org/patch/3184/>
Is there a better example for me to use?
inters on what I'm doing wrong?
Thanks,
Ronak
private func filterBuffer(_ inputBufferList:
UnsafeMutableAudioBufferListPointer, frameCount: UInt32, outputBufferList:
UnsafeMutableAudioBufferListPointer) throws -> UInt32 {
guard let inputAudioFrame = inputAudioFrame, let outputAudio
#x27;t the best for FFmpeg.
Thanks,
Ronak
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> On Dec 20, 2018, at 1:03 AM, Taner Sener wrote:
>
> On Wed, 19 Dec 2018 at 19:08, Ronak
> wrote:
>>
>> Therefore, I'm trying to find out the best way to convert the Ffmpeg
> .dylibs to .frameworks.
>>
>> I tried to just package the .dylibs (rena
> On Dec 19, 2018, at 11:15 AM, Carl Eugen Hoyos wrote:
>
> 2018-12-19 17:08 GMT+01:00, Ronak :
>
>> I'm trying to use Ffmpeg inside of an Apple framework; (built for all of
>> Apple's platforms). Unfortunately; I'm learning that Apple does not
support building as .frameworks from the get go?
Thanks,
Ronak
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> 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, 201
> 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, 20
> 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
> 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
> On Dec 10, 2018, at 7:29 PM, Carl Eugen Hoyos wrote:
>
> 2018-12-11 1:07 GMT+01:00, Ronak :
>> Hey guys,
>>
>> I'm trying to use the dynaudnorm and earwax filters in my iPhone app, and
>> I've noticed that the dynaudnorm filter wants to resample
> On Nov 27, 2018, at 10:46 AM, Carl Eugen Hoyos wrote:
>
> 2018-11-12 18:20 GMT+01:00, Ronak :
>
>> I'm trying to get Ffmpeg to package and generate a valid fMP4 file
>> for HLS of an Audio Only Dolby Digital Plus file.
>
> Is your input file really an a
lCount)))
frame?.pointee.sample_rate = Int32(format.sampleRate)
frame?.pointee.format = Int32(AV_SAMPLE_FMT_FLTP.rawValue)
frame?.pointee.nb_samples = Int32(maximumFrameCount)
What am I doing wrong?
Thanks,
Ronak
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Hey Carl,
I figured out the correct fix for this, based on your suggestion.
I'm going to send a patch for this to the dev mailing list. Appreciate your
help to review my change.
Ronak
> On Nov 16, 2018, at 2:16 PM, Carl Eugen Hoyos wrote:
>
> 2018-11-16 0:46 GMT+01:00, Ronak :
n Nov 12, 2018, at 12:11 PM, Ronak wrote:
>
> Hey All,
>
> I noticed that ffmpeg always seems to generate a SIDX box when it's not
> really required for Audio Only content.
>
> Example command:
>
> ffmpeg -i atmosTest.mp4 -codec copy -hls_time 0.99
L C R Ls Rs LFE)
I tried examining the generated M4S file from FFMPEG, and I noticed that the
sample tables in the tkhd atoms are all 0s. Ffmpeg is not able to properly
encode the audio into the fragments.
How can I go about debugging this is
21 kbits/sec - max file bit
rate is 768.24 kbits/sec
It generates a file without any SIDX atoms; only ftyp, moov & a collection of
moof & mdat atoms.
Why does ffmpeg do this? How hard would it be to remove this behavior from
ffmpeg?
Thanks,
Ronak
_
Thanks.
Do you know when this would be merged in so I can build the latest developer
trunk?
There are features in the developer trunk that I'm interested in using.
> On Oct 27, 2018, at 3:32 PM, Mark Thompson wrote:
>
> On 27/10/18 17:24, Ronak wrote:
>> Hi all,
>>
lude/linux/videodev2.h.
What do I have to set to make compilation succeed?
Thanks,
Ronak
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I figured out how to do this. The problem was just that I forgot to put the
-map 0:0 -map 1:0, etc. arguments in these commands. So ffmpeg was only
selecting the first stream in the input.
> On Aug 1, 2018, at 12:03 PM, Ronak wrote:
>
> Hi all,
>
> I'd like to generate
ine I should be using for this? If this doesn't
work, I'll file an issue to ffmpeg for this, and try to figure out how to fix
this.
Thanks,
Ronak
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> On Jul 12, 2018, at 6:21 PM, Ronak Patel
> wrote:
>
> Hey Carl,
>
> So I dug into this more today and I have root caused what's exactly happening
> here.
>
> The problematic code is this:
> https://github.com/FFmpeg/FFmpeg/blob/master/libavformat/hl
something like this, looks like we'd have to significantly
re-engineer the code. Do you have any pointers on how to go about doing this?
Or, would you be able to help do this?
Thanks for all your help,
Ronak
> On Jun 27, 2018, at 2:04 PM, Carl Zwanzig wrote:
>
> Hi,
>
> I h
do this today? Or is this an enhancement I’d have to file? Or
figure out how to add myself?
Thanks
Ronak
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> On Jun 27, 2018, at 1:50 AM, Carl Zwanzig wrote:
>
> On 6/26/2018 9:20 PM, Ronak wrote:
>> The temp files are made in the CWD. And they are constantly created and
>> destroyed for every single fragment it seems. I looked into the ffmpeg
>> code and it looks like ev
> On Jun 25, 2018, at 6:03 PM, Carl Zwanzig wrote:
>
> On 6/25/2018 2:41 PM, Carl Eugen Hoyos wrote:
>> 2018-06-25 15:02 GMT+02:00, Ronak:
>>> 1. Apple does not create temp files
>> Isn't that generally a disadvantage?
>
> Why would it be (not creat
Hi,
I was curious if ffmpeg has plans to support hlsv8 generation.
The ability to do variable substitution in manifest is huge and can really help
reduce manifest sizes.
Ronak
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> On Jun 23, 2018, at 12:07 AM, Ronak wrote:
>
>
>
>> On Jun 22, 2018, at 7:59 PM, Ronak Patel wrote:
>>
>>
>>> On Jun 22, 2018, at 5:03 PM, Carl Zwanzig wrote:
>>>
>>>> On 6/22/2018 1:37 PM, Ronak wrote:
>>>>
> On Jun 22, 2018, at 4:45 PM, Ronak wrote:
>
>
>
>>> Hi,
>>>
>>> We are trying to setup an adaptive stream across two heaac streams and
>>> higher order lcaac ones.
>>>
>>> So we’d have something like so:
>>>
&
> On Jun 22, 2018, at 7:59 PM, Ronak Patel wrote:
>
>
>> On Jun 22, 2018, at 5:03 PM, Carl Zwanzig wrote:
>>
>>> On 6/22/2018 1:37 PM, Ronak wrote:
>>> We have audio files that are more than 100 hours long, and we need them
>>> to be fragment
Sent from my iPhone
> On Jun 22, 2018, at 5:03 PM, Carl Zwanzig wrote:
>
>> On 6/22/2018 1:37 PM, Ronak wrote:
>> We have audio files that are more than 100 hours long, and we need them
>> to be fragmented quickly. It totally seems like there's an I/O problem
>
> On Jun 22, 2018, at 4:31 PM, Ronak wrote:
>
>
>
>> On Jun 22, 2018, at 3:59 PM, David Favor wrote:
>>
>> I'd say you're looking at a very long transcode time,
>> as your video dump shows...
>>
>> Duration: 52:15:29.84, star
a problem with how ffmpeg is
> generating the HEAACv2 fmp4 files?
>
>
Could anyone please shed any light on why the defaultSampleDurations in moof
are different between LC & HE-AAC at the same sampling rates?
>> Thanks,
>>
>> Ronak
>> __
> On Jun 22, 2018, at 3:59 PM, David Favor wrote:
>
> I'd say you're looking at a very long transcode time,
> as your video dump shows...
>
> Duration: 52:15:29.84, start: 0.00, bitrate: 63 kb/s
>
> So... a 52 hour video... well... better fire up the popcorn
> maker + settle in for a ve
> On Jun 20, 2018, at 4:48 PM, Carl Eugen Hoyos wrote:
>
> 2018-06-20 18:37 GMT+02:00, Ronak :
>
>> That command took the following time:
>>
>> real 3m19.316s
>> user 3m14.730s
>> sys 0m1.830s
>>
>> I'm curious about your thoug
> On Jun 21, 2018, at 9:02 AM, Ronak Patel
> wrote:
>
> Hi,
>
> We are trying to setup an adaptive stream across two heaac streams and higher
> order lcaac ones.
>
> So we’d have something like so:
>
> HEAACv2 44khz/32kbps
> HEAACv2 44/64
> LCA
Sorry I realized I sent you an mpeg ts example below, but I’ve seen the same
with fmp4. This tells me that the logic that decides how to fragment the
segments is independent of the file format.
Sent from my iPhone
> On Jun 21, 2018, at 9:09 AM, Ronak Patel
> wrote:
>
> Hi,
&g
ile format? Was this fixed in newer versions?
Thanks,
Ronak
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pling rate of the LC-AAC audio?
Thanks,
Ronak
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What do you mean by top-post? I'm just responding to the email chain through
the mailing list. What's the correct etiquette?
> On Jun 20, 2018, at 4:48 PM, Carl Eugen Hoyos wrote:
>
> 2018-06-20 18:37 GMT+02:00, Ronak :
>
>> That command took the following time:
&
it here? Either in writing more
logs or not buffering enough content in RAM to write out to disk in larger
bursts.
Ronak
> On Jun 20, 2018, at 12:12 PM, Carl Eugen Hoyos wrote:
>
> 2018-06-20 16:51 GMT+02:00, Ronak :
>
>> Ffmpeg -I "${FILE}.mp4" -codec copy -f
22c3e80] Opening 'bk_acx0_006556_22_32_1sec.mpd.tmp' for writing
[dash @ 0x22c3e80] Opening 'media_0.m3u8.tmp' for writing
[dash @ 0x22c3e80] Opening 'bk_acx0_006556_22_32_1sec.m4s' for reading
[dash @ 0x22c3e80] Opening 'bk_acx0_006556_22_32_1sec.mpd.tmp' for w
is?
Here's the command we're running: ffmpeg -i "${FILE}.mp4" -codec copy -hls_time
9.75238095238095 -hls_segment_type fmp4 -hls_flags single_file+append_list
-hls_playlist_type vod "${FILE}_10sec_v7.m3u8"
There's no encoding here
Hi Carl,
What do you mean by "just test with ffmpeg?"
What option prints this information out? Or what code would I have to write?
Ronak
> On Jun 12, 2018, at 4:14 AM, Carl Eugen Hoyos wrote:
>
> 2018-06-12 2:14 GMT+02:00, Ronak Patel :
>
>> I was curious abo
, what would be the best way to
calculate this?
Thanks
Ronak
Sent from my iPhone
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, what would be the best way to
calculate this?
Thanks
Ronak
Sent from my iPhone
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EG DASH. Is there an ETA
on when this would be available in ffmpeg's trunk?
Thanks,
Ronak
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maybe one of them isn't consistent?
Ronak
> On Mar 17, 2018, at 8:02 AM, Carl Eugen Hoyos wrote:
>
> 2018-03-17 12:38 GMT+01:00, Ronak Patel :
>> Hello,
>>
>> I’m encoding my wav files to HE AAC v2 in 44kHz/32 Kbps and 44/64.
>
> Command line and compl
/3971
Thanks
Ronak
Sent from my iPhone
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inux?
Thanks,
Ronak
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