[Libav-user] OPUS transcoding to AAC but 960 sample increase to 1024 with a nomalized blank data.

[He Lei]

Sorry, I didn't make myself clear.

============================
while(has_data){

    get_960_samples();

    av_audio_fifo_write(960);

    while(avio_audio_fifo_size() >= 1024){

      avio_fifo_read(1024);

      encode(1024); //encode in hear or other

    }
}

if(avio_audio_fifo_size() % 1024){

   fill_mute(1024 - avio_audio_fifo_size());  //Only fill the mute at the last!

   avio_fifo_read(1024);

   encode(); //encode in hear or other
}
============================


LeiHe
helei0908 at hotmail.com<mailto:helei0908 at hotmail.com>



在 2018年10月29日，下午12:52，강구철 <kckang at skycom.ne.kr<mailto:kckang at skycom.ne.kr>> 写道：

I already tried using av_fifo for make it size to 1024. but still encoding api output 960 plus blank 68 samples(total 1024).

//AVCodecContext initialization
AVCodecContext  *opus_context;
AVCodecContext  *aac_context;

        opus_context->channels = 1;
        opus_context->channel_layout = AV_CH_LAYOUT_MONO;
        opus_context->sample_rate    = 48000;//SRATE;
        opus_context->sample_fmt     = AV_SAMPLE_FMT_FLTP;//4BYTE
        opus_context->bit_rate       = 48000;//BITRATE;

        aac_context->channels = 1;
        aac_context->channel_layout = AV_CH_LAYOUT_MONO;
        aac_context->sample_rate    = 48000;//SRATE;
        aac_context->sample_fmt     = AV_SAMPLE_FMT_FLTP;
        //aac_context->sample_fmt   = AV_SAMPLE_FMT_S32;
        aac_context->bit_rate       = 48000;//BITRATE;//64000
                     //aac_context->strict_std_compliance = FF_COMPLIANCE_EXPERIMENTAL;

        fifo = av_audio_fifo_alloc(aac_context->sample_fmt, aac_context->channels, 1);

        opus_codec = avcodec_find_decoder( AV_CODEC_ID_OPUS );
        aac_codec  = avcodec_find_encoder( AV_CODEC_ID_AAC  );

//transcoder function for trans OPUS => ACC

tcode(AVPacket* src, AVPacket* dst){
           opus_context->frame_size=960;//TESTTEST
           decoded_frame->data[0] = (uint8_t*)av_malloc(4*1024);//TESTTEST
           ret = avcodec_decode_audio4(opus_context, decoded_frame,  &data_present, src); // decoded_frame 960 sucess
}



    //WAVE Generate for 68 samples------------------------------------------------------------
    unsigned char _cbuf[100];
    unsigned char *cbuf=NULL;
    cbuf = _cbuf;
    cbuf[0]=0xf2;
    cbuf[1]=0xdb;
    cbuf[2]=0x0;
    cbuf[3]=0x3f;
    decoded_frame->nb_samples = 1024;//change to 1024 for AAC encoding block

    for(int i=960; i<1024 ; i++){
               decoded_frame->data[0][i*4+0] = (uint8_t)cbuf[0]; //float
               decoded_frame->data[0][i*4+1] = (uint8_t)cbuf[1]; //float
               decoded_frame->data[0][i*4+2] = (uint8_t)cbuf[2]; //float
               decoded_frame->data[0][i*4+3] = (uint8_t)cbuf[3]; //float
               cbuf[2] += 2;        //make saw type wav using range is 0~96
               if(cbuf[2] >= 96)cbuf[2]=0;//mod reset to 0
    }//END OF Generate 68 sample of last WAVE(PCM Float 4byte LittleEndian type)---------------------------------------------------------

ret = avcodec_encode_audio2(aac_context, dst, decoded_frame, &data_present);//AAC ENCODING Success 1024 sampes compressed but contain last 68 samples blank.

this mail attached sample result aac before muxing and next figure show you every 20msec data has 64 nomalized blank samples(marked red pen). thanks for any recommand or inform.

<abc1.m4a AAC recorded webrtc sample open with audacity>
<image001.png>

From: Libav-user [mailto:libav-user-bounces at ffmpeg.org] On Behalf Of He Lei
Sent: Friday, October 26, 2018 6:16 PM
To: This list is about using libavcodec, libavformat, libavutil, libavdevice and libavfilter.
Subject: Re: [Libav-user] OPUS transcoding to AAC but 960 sample increase to 1024 with a nomalized blank data.

Using “audio_fifo” to cache samples, When the samples number in fifo is enough to 1024, and then encode it.
The last, If the number of samples  is less than 1024 in fifo, fill with mute


look at “doc/examples/transcode_aac.c”

LeiHec
helei0908 at hotmail.com<mailto:helei0908 at hotmail.com>




在 2018年10月26日，下午4:34，강구철 <kckang at skycom.ne.kr<mailto:kckang at skycom.ne.kr>> 写道：

Im transcode voice comming from WebRTC through by RTP with h264 video.
received sound unit is 20msec OPUS stereo 48000 2channel sample per second
its good decoded to PCM32 FLTP type and good play.
after decode I encode to AAC 48000 stereo frame nb_samples is 960. but encoding ffmpeg aac function
attach 64 samples every each decoded raw PCM samples. what should I do for it to improving final aac product quality ?

now I found AAC Context be able to control cypher block size 1024 to 960. some documents say aac encoder default block is 1024.

AACContext *ac = (AACContext*)aac_context->priv_data;
MPEG4AudioConfig *m4ac = &(ac->oc[0].m4ac);
m4ac->frame_length_short = 1;//1:960, 0:1024

is this right approching ? appriciate any kinds of oppinion of you guys!!


_______________________________________________
Libav-user mailing list
Libav-user at ffmpeg.org<mailto:Libav-user at ffmpeg.org>
http://ffmpeg.org/mailman/listinfo/libav-user

<abc1.m4a>_______________________________________________
Libav-user mailing list
Libav-user at ffmpeg.org<mailto:Libav-user at ffmpeg.org>
http://ffmpeg.org/mailman/listinfo/libav-user

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://ffmpeg.org/pipermail/libav-user/attachments/20181029/28c51e01/attachment.html>