Dear all,
I am getting stuck with decoding some audio files that include many audio
streams ( more than 1 ). Even though i used av_find_best_stream(formatContext,
AVMEDIA_TYPE_AUDIO, -1, -1, NULL, 0) function, the result is so bad. It can
just decode very few frames and stop. My audio files can
Hi,
I've noticed something which I think it's a strange behavior of ffmpeg
command line.
Encoding the same file, once treating it as a wav, 2nd time as a raw file
(simply removing the wav header), it behaves differently.
Case 1 - encoding a wav as input (44wav head plus the actual PCM data):
./ff
Hi,
I am implementing ffmpeg as a mex function in matlab as part of a research
project (with main focus on the audio, not video like OpenCV, etc).
I managed to have an initial version of audio encoding working, but I need
to clarify few things before considering it usable by others with other
cod
ReSearchIT Eng writes:
> ./ffmpeg.exe -loglevel debug -y -vn -f s16le -ac 1
> -ar 16000 -acodec pcm_s16le -sample_fmt s16 -i 684.wav
Your questions are difficult to understand...
Lets start with the simple things:
Please remove everything starting with "-f" until "s16":
If "ffmpeg -i 684.wav"
On Tue, 8 Apr 2014 15:28:05 +0300
ReSearchIT Eng wrote:
> Hi,
>
> I am implementing ffmpeg as a mex function in matlab as part of a research
> project (with main focus on the audio, not video like OpenCV, etc).
>
> I managed to have an initial version of audio encoding working, but I need
> to
ReSearchIT Eng writes:
> the input or the output format?
As said, this is difficult to understand:
FFmpeg / libavcodec support decoding audio
(to raw audio, different formats depending
on the input) and encoding audio (from raw
audio, format depending on the encoder you
requested, unfortunat
Hi Eugen,
Thanks for the quick reply.
For case 1(wav), I tried now again removing the parameters as suggested.
Removing them does not make any change (see below).
Note: I came to the long set of parameters, in order to have the two
commands as close as possible, to leave no "doubts".
*Case 1: (WA
ReSearchIT Eng writes:
> I tried now again removing the parameters as suggested.
> Removing them does not make any change (see below).
This is (as such) impossible, you are now encoding
44 bytes less (which were not audio).
Please:
Do not post excerpts from ffmpeg's console output,
always po
ReSearchIT Eng writes:
> Command: ./ffmpeg.exe -loglevel debug -y -vn -f s16le
> -ac 1 -ar 16000 -acodec pcm_s16le -sample_fmt s16
> -i 684.wav -ac 1 -ab 23850 -ar 16000 -f amr
> -acodec libvo_amrwbenc -sample_fmt s16
> 640bytes_fromwav.awb
Please remove:
"-acodec pcm_s16le", this is the onl
Hi everyone,
I am going mad trying to find out where is the memory leak in a piece of
code I am using in my project! I have a list of AVPacket structs and I do
want to put them in a mp4 file. I successfully do it but in each iteration
memory increases about 400 kBytes. To ensure that the problem i
Don't know what it means by top-post, just replying from gmail interface.
"-f amr -acodec libvo..." -> Itried to remove, but is a must to keep
them, it does not work without both of them. (adding only -f amr, it
defaults to amrb instead of amrwb; adding only -acodec lib.. -> gives error
it does
Hi,
I've noticed something which I think it's a strange behavior of ffmpeg
command line.
Encoding the same file, once treating it as a wav, 2nd time as a raw file
(simply removing the wav header), it behaves differently.
Case 1 - encoding a wav as input (44wav head plus the actual PCM data):
./ff
Actually I am trying ffmpeg in ubuntu but getting exception that it is
depricated and use libav. Sometimes ffmpeg command doesn't work on ubuntu,
with warning message. Therefore I am using libav. Kindly suggest ow to make
ffmpeg compatible with ubuntu.
--
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I'm taking a look at it, thanks. But, just to be clear, this isn't possible
with ffmpeg?
2014-04-08 1:20 GMT-03:00 Tocy :
> recommend you to refer jsvm ,SVC for H.264/AVC
>
>
> -- Original --
> *From: * "Diego Barboza";;
> *Date: * Mon, Apr 7, 2014 09:24 PM
> *To:
On 07/04/14 16:49, Gonzalo Garramuno wrote:
I am adding muxing to my video player and I am running into a wall.
If I use mp3 (libmp3lame) as the audio encoding method, the resulting
file has perfect audio that ends up short (like 256 frames). If I use
PCM_S16LE, the audio is complete albeit I
Hi
I want to configure ffmpeg to enable only one encoder but disable all rest
encoders.
I tried this way ./configure --disable-encoders --enable-encoder=aac
seems it doesn't work.
So is there any way to configure without explicitly listing all encoder
names
Thanks
_
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