[Libav-user] AAC encoding question
justin-zhao at qq.com
Sun Apr 7 08:45:18 CEST 2013
Thank you firstly, I have tried all sample format by for loop, but it still didn't work.
------------------ Original ------------------
From: "Haridas Sagar N"<haridassagarn at tataelxsi.co.in>;
Date: Sun, Apr 7, 2013 02:40 PM
To: "This list is about using libavcodec, libavformat, libavutil, libavdeviceand libavfilter."<libav-user at ffmpeg.org>;
Subject: Re: [Libav-user] AAC encoding question
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
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!
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");
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