nd 'probesize'
options
\\audio_out.pcm: could not find codec parameters
Input #0, aac, from '\\audio_out.pcm':
Duration: 00:00:11.94, bitrate: 1535 kb/s
Stream #0:0: Audio: aac, 4.0, fltp, 1535 kb/s
At least one output file must be specified
On Mon, Jan 23, 2017 at 8:04 PM, Steve Myers <musicspee
> On Jan 23, 2017, at 5:38 PM, Cesareo Fernandez wrote:
>
> I have been attempting to use the ffmpeg libraries to take as input a RTMP
> stream consisting of a video stream and an audio stream, and decode the
> respective streams in order to push the contents out to a
Due to circumstances I lost the build script I used to build the shared
libraries for my app. Is there a way a to get the configuration used from
shared libraries similar to the command line ffmpeg? I only have
libavcodec-57.so, libavformat-57.so, etc.
It's shorts in memory that must be processed in real time.
On Dec 1, 2016 6:46 PM, "Carl Eugen Hoyos" <ceffm...@gmail.com> wrote:
> 2016-12-01 21:13 GMT+01:00 Steve Myers <musicspeedchan...@gmail.com>:
> > Looking at the filtering example, it appears
Looking at the filtering example, it appears AVFrame's are needed to pass
to the filter graph. Is it possible/feasible to create our own AVFrame's if
we have raw PCM audio of a known format in order to pass to the filters? Is
there any way else? Somewhere that shows an example of such thing?
It is possible to compile for armeabi, armeabi-v7 neon and non-neon as
well. Also you can compile a version for arm64 too. However I don't think
there are any Android anymore that are not v7 compatible, and non-neon is
almost non existent as well.
So a armeabi-v7 neon build would be compatible
Sorry
I think
---enable-demuxer=mp3
--enable-decoder=mp1,mp2,mp3,mp3adu
On Wed, Oct 26, 2016 at 11:00 AM, Steve Myers <musicspeedchan...@gmail.com>
wrote:
> I think
>
> ---enable-demuxer=mp3
> --enable-decoder=mp1,mp2,mp3adu
>
> On Wed, Oct 26, 2016 at 10:53 AM,
I think
---enable-demuxer=mp3
--enable-decoder=mp1,mp2,mp3adu
On Wed, Oct 26, 2016 at 10:53 AM, Carl Eugen Hoyos
wrote:
> 2016-10-26 14:02 GMT+02:00 Diaz Soho :
>
> > when I enable decoder "mp3", the mpegaudio does not enabled.
> > And in advance, I
Do you have demuxer and parser too?
On Oct 26, 2016 8:10 AM, "Diaz Soho" wrote:
> Hi all,
>
> when I enable decoder "mp3", the mpegaudio does not enabled. And in
> advance, I check that it seems dependence DCT. I do not set
> "--disable-dct", the CONFIG_DCT is 0 always.
On Tue, Oct 25, 2016 at 10:32 AM, Jason C wrote:
> In 3.1.5 it appears that AVStream::codec is deprecated:
>
> /**
> * @deprecated use the codecpar struct instead
> */
> attribute_deprecated
> AVCodecContext *codec;
>
> But the examples linked to
Yeah I'm rolling it back 2048 samples (arbitrary) from where I want to seek
to and then dropping those extra samples and it is working now. Still odd
though.
On Sep 11, 2016 6:24 PM, "Christopher Snowhill" wrote:
>
>
> On 9/11/16 12:30 PM, Steve wrote:
> > Steve wrote
> >>
I'm using ffmpeg audio decoding to create a gapless audio loop. I call
av_seek_frame (with AVSEEK_FLAG_BACKWARD) to seek to my desired time, and
call avcode_flush_buffers after. It seems the first decoded frame in the
audio stream after the seek contains a few samples of silence or near
silence,
I'm trying to build a small universal audio decoding library that decodes
the file a frame at a time (on Android platform). I've built a command line
static ffmpeg that I can exec that works on every file format I have
tested. Now I am trying to build a shared library version, so I can
directly
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