Hello,
Hey sorry. I got it. I forgot to set codec_id in AVCodecContext. Thanks for the
reply.
I had one more doubt, the output file after this change, plays in vlc but can't
hear any sound. When I checked the meta info on the file, the bits per sample
(bit depth) is 32. Is that correct for libopus ? If not where can I set the bit
depth in FFMpeg so that it considers correct no of bits per sample ? I guess
the problem is in that.
And, one more thing is AVCodecContext.time_base I am setting it as { 1, 48000}
. Actually, it is not mentioned in example docs to set time base for audio
codec. But I read somewhere we have to set it, thats why I am setting it. Is it
correct ?
Thanks.
Regards,
Ravi Kiran B S
---- On Fri, 16 Feb 2018 20:32:36 +0530 ravikiran.bs
<[email protected]> wrote ----
Hello,
Thanks for reply. You mean, codec_id in
AVFormatContext->oformat->audio_codec ?? I am setting it.
Regards,
Ravi Kiran B S
---- On Fri, 16 Feb 2018 20:14:08 +0530 Carl Eugen Hoyos
<[email protected]> wrote ----
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2018-02-15 15:47 GMT+01:00 ravikiran.bs < [email protected]>:
> I am trying to mux a opus audio stream to a webm file. (I know audio only
> can be muxed into a ogg container. I tried that. But, FFMpeg is not
> detecting the file format when I write the file headers using
> avformat_write_header() of libavformat). I have configured the FFMpeg
build
> with --enable-libopus and --enable-libvorbis. Now, after writing the audio
> data when I call av_write_trailer() to close the file, it is throwing the
> following error :
>
> "Only VP8 or VP9 video and Vorbis or Opus audio and WebVTT subtitles are
> supported for WebM."
It appears you did not set codec_id to AV_CODEC_ID_OPUS.
Carl Eugen
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