On Thu, Apr 28, 2016 at 02:09:21PM +, JULIAN GARDNER wrote:
> Ok added the code from filtering_audio.c to create a filterchain, changed my
> code to take the decoded frames and push into the filter.
> filt_frame is returned and this is pushed into the avcodec_encode_audio2 and
> guess what
Ok added the code from filtering_audio.c to create a filterchain, changed my
code to take the decoded frames and push into the filter.
filt_frame is returned and this is pushed into the avcodec_encode_audio2 and
guess what error i get
[mp2 @ xxx] nb_samples (1024) != frame_size (1152)
On Thu, Apr 28, 2016 at 01:00:37PM +, JULIAN GARDNER wrote:
> So my question is do I have to split the input to be the correct size or am i
> missing something in the audio encoding setup to allow it to do the buffering
> correctly?
Apparently yes.
I believe libavfilter API can do both
I am trying to get my code to work and failing miserably.
I have a file opened which has an audio stream in AC3, 48k, Stereo, fltp,
192kb/s
I am trying to output this as MP2, 64k, Stereo, S16, 32 kb/s
Now my code goes through the svr_convert and gives me a 1040 sample buffer from
the 1536 input
On Thu, Apr 28, 2016 at 04:54:27PM +0800, qw wrote:
> Hi,
>
> I define the following filter graph:
>
> setpts=PTS-STARTPTS, fps=fps=20.00, scale=500:280;
> drawtext="fontfile=/usr/share/fonts/truetype/freefont/FreeSerif.ttf:
> text='Test Text'"
> And avfilter_graph_parse_ptr(), and
Hi,
I define the following filter graph:
setpts=PTS-STARTPTS, fps=fps=20.00, scale=500:280;
drawtext="fontfile=/usr/share/fonts/truetype/freefont/FreeSerif.ttf: text='Test
Text'"
And avfilter_graph_parse_ptr(), and avfilter_graph_config() are used to create
filter graph.
How to set the name
