[FFmpeg-user] Realtime Audio Transcoding delay

[Roger Pack]

On 12/21/12, shahab shirazi <shahab.sh.70 at gmail.com> wrote:
> Hi,
>
> I'm writing an Android app. This app gets audio stream from the mic,
> encodes it and sends this stream to my server (through a tcp connection).
> The server is supposed to read this stream, transcode it on the fly and
> send it to an external device which only supports g726 audio format.
>
> Now the problem is the delay. Currently it's about 10 seconds. After some
> digging here is what I found:
> The Android app -> server seems to be working fine. Sends audio stream of
> AMR-NB format, 8000 Hz, 1 channels, flt, 12 kb/s
> Server feeds ffmpeg (or vlc) at about 1.6 KBytes/s and it takes about 8
> seconds for ffmpeg/vlc to output any data. This is the bottle neck.
>
> Here is the command I used for ffmpeg:
>     ffmpeg -i - -acodec g726 -ar 8k -ac 1 -b:a 8k -f wav -

Is there latency introduced "before" it hits ffmpeg?