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<DIV>Hi,</DIV>
<DIV> </DIV>
<DIV>I have the same issue.</DIV>
<DIV> </DIV>
<DIV>At the beginning, aac encoder doesn't output coded data for the first two
frames, which means encoder begins to output coded data from 3rd frame. In the
end, all audio samples are sent to aac encoder, but coded data of last
samples are not output yet.</DIV>
<DIV> </DIV>
<DIV>I know how to flush delayed frames in h264 encoder, i.e. flush operation
uses avcodec_encode_video()/avcodec_encode_video2() like encoding operation. But
avcodec_encode_audio()/avcodec_encode_audio2() can't flushed delayed
samples in aac encoder, as aac encoder doesn't have
the AV_CODEC_CAP_DELAY capability set.</DIV>
<DIV> </DIV>
<DIV>'int avcodec_encode_audio2( AVCodecContext *avctx, AVPacket *avpkt, const AVFrame *frame, int *got_packet_ptr )
' is the function of encoding a audio frame. If 3rd argument frame is set to
NULL, aac encoder won't flush delayed samples, as aac encoder doesn't have
the AV_CODEC_CAP_DELAY capability set.</DIV>
<DIV> </DIV>
<DIV>Which ffmpeg function can flush delayed aac samples? How to flush
those samples?</DIV>
<DIV> </DIV>
<DIV>I have tried one way to flush those samples. Having sent all audio frames
to aac encoder, I continue send to aac encoder several frames which have
samples with zero value. AAC does send coded data of delayed samples, but I
don't think it is good to flush aac encoder.</DIV>
<DIV> </DIV>
<DIV>Is there better way to flush aac encoder?</DIV>
<DIV> </DIV>
<DIV>andrew</DIV>
<DIV> </DIV>
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<DIV><B>From:</B> <A href="mailto:panicosk@gmail.com">Panicos
Karkallis</A></DIV>
<DIV><B>Date:</B> 2015-08-22 00:44</DIV>
<DIV><B>To:</B> <A href="mailto:libav-user@ffmpeg.org">This list is about
using libavcodec, libavformat, libavutil, libavdevice and libavfilter.</A></DIV>
<DIV><B>Subject:</B> Re: [Libav-user] AAC encoder not returning final
samples</DIV></DIV></DIV>
<DIV>
<DIV class=FoxDiv20150822011210349497>
<P dir=ltr>I do flush the encoder by passing a null frame. it returns 2 packets
that get appended at the beginning of my file and increase the padding instead
of giving the end samples.</P>
<P dir=ltr>Panicos</P><BR>
<DIV class=gmail_quote>
<DIV dir=ltr>On 17:34, Fri, 21 Aug 2015 Henry, David <<A
href="mailto:David.Henry@echostar.com"
target=_blank>David.Henry@echostar.com</A>> wrote:<BR></DIV>
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<DIV>
<DIV>
<DIV>> I am new to ffmpeg/avlib and what I am trying to do is just decode
and encode back again an aac file. I don't try to make any changes to the file
although the avlib encoder is producing a file with more initial padding than
expected and it is cropping the final frames of the audio
clip.</DIV></DIV></DIV><SPAN>
<DIV dir=ltr>
<DIV>> I exported the avframes created by the decoder and all the file
information is there. It is at the encoding step that I am doing something
wrong.</DIV></DIV></SPAN>
<DIV><BR></DIV></DIV>
<DIV
style="WORD-WRAP: break-word; FONT-SIZE: 14px; FONT-FAMILY: Calibri,sans-serif; COLOR: rgb(0,0,0)"><SPAN>
<DIV dir=ltr>
<DIV>It sounds like you are not flushing the libfdk-aac encoder at the
end. The encoder holds on to audio samples internally while you are
encoding and you need to flush those last samples</DIV></DIV></SPAN>
<DIV>When you are finished encoding.</DIV>
<DIV>-
David</DIV></DIV>_______________________________________________<BR>Libav-user
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