[Libav-user] AAC encoding question

Haridas Sagar N haridassagarn at tataelxsi.co.in
Sun Apr 7 08:40:29 CEST 2013

Change sample format
From: libav-user-bounces at ffmpeg.org [libav-user-bounces at ffmpeg.org] on behalf of Justin [justin-zhao at qq.com]
Sent: Sunday, April 07, 2013 11:38 AM
To: libav-user
Subject: [Libav-user] AAC encoding question


    I am trying to encode audio using the AAC audio encoder with the program below, but when I call the avcodec_open2 funtion, the function always return -733130664. I don't know where is wrong in it. I'll very appreciate someone who can point out the wrong.

note:if the codec AV_CODEC_ID_AAC is changed as AV_CODEC_ID_MP2, the avcodec_open2 can return 0 successfully!

 int ret;
 AVCodec *codec;
 AVCodecContext *c;
 AVFrame *frame;

 codec = avcodec_find_encoder(AV_CODEC_ID_AAC);
 if (!codec)  return;
 c = avcodec_alloc_context3(codec);
 if (!c)  return;

 c->bit_rate = 64000;
 c->sample_rate    = 44100;
 c->channels       = 2;
 c->frame_size = 88200;
 c->sample_fmt = AV_SAMPLE_FMT_S16;
 // open it
 ret = avcodec_open2(c, codec, NULL);
 if (ret >= 0)  printf("OK\r\n");

Best Regards,
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