I don' t know if this is your problem or not, but I think you can
replace all malloc() and free() functions with av_malloc() and
av_free(). My understanding is that the av versions ensure that the
memory is aligned properly so you don't need to use posix_memalign().
On Sun, Jul 17, 2011 at 4:53 PM, alex b. <ab...@avail-tvn.com> wrote:
>
>
> Hi,
>
> I am trying to encode audio using the AAC audio encoder with the program
> below. It seems to work fine (runs without issues) but the resulting AAC
> audio is not playable with anything like ffplay or mplayer. If anyone can
> point out what I'm doing wrong that would be really appreciated.
>
> The input.wav file is made up of 2 channel 6144 byte samples, 5 to each
> frame (for a frame size of 30720 byes) and it plays fine in mplayer with a
> command like
>
> $ mplayer -demuxer rawaudio -rawaudio rate=48000 input.wav
>
> there is no header in the .wav file, but that's ok. also, encoding this
> input using ffmpeg and the "-strict experimental" option for AAC audio works
> fine as well
>
>
> thanks, alez...
>
>
> -----------------------------------------------------------
>
> /* test audio encoder */
> #define _XOPEN_SOURCE 600
>
> #include <stdint.h>
> #include <stdlib.h>
> #include <string.h>
> #include <inttypes.h>
> #include <stdio.h>
> #include <avcodec.h>
> #include <libavcodec/opt.h>
> #include <libavutil/log.h>
>
> #define FRM_SZ 30720
> #define SMPL_SZ 6144
>
> int main(void)
> {
> AVCodec *codec;
> AVCodecContext *context;
> int bytes_read;
> FILE* in_pcm = fopen("input.wav", "rb");
> FILE* out_aac = fopen("output.aac", "wb");
>
> uint8_t* inbuff = malloc(FRM_SZ*sizeof(uint8_t));
> uint8_t* smallbuff;
>
> avcodec_init();
> avcodec_register_all();
>
> codec = avcodec_find_encoder(CODEC_ID_AAC);
>
> context = avcodec_alloc_context3(codec);
>
> context->bit_rate = 128000;
> context->sample_rate = 48000;
> context->channels = 2;
> context->frame_size = 30720;
> context->sample_fmt = AV_SAMPLE_FMT_S16;
>
> if (avcodec_open2(context, codec, NULL) < 0)
> {
> fprintf(stderr, "ERROR: could not initialize encoder\n");
> exit(1);
> }
>
> while ( ( bytes_read = fread( inbuff, FRM_SZ, 1, in_pcm ) ) > 0 )
> {
> // there are 5 audio samples per frame (4620 frames total = 28385280
> bytes in the file)
> for ( int i = 0; i < 5; i++ )
> {
> smallbuff = malloc(SMPL_SZ*sizeof(uint8_t));
>
> memcpy(smallbuff, &inbuff[i*SMPL_SZ], SMPL_SZ);
>
> int frame_bytes = context->frame_size * context->sample_fmt *
> context->channels;
> int outbuf_size;
>
> uint8_t* outbuf;
> posix_memalign( (void**)&outbuf, (size_t)16, (size_t)frame_bytes
> );
>
> outbuf_size = avcodec_encode_audio(context, outbuf, frame_bytes,
> (short*)smallbuff);
>
> fwrite(outbuf, outbuf_size, 1, out_aac);
>
> free(outbuf);
> free(smallbuff);
> }
> }
>
> free(inbuff);
> return 0;
> }
>
>
>
> _______________________________________________
> Libav-user mailing list
> Libav-user@ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/libav-user
>
--
-
-
-
-
Luke Clemens
http://clemens.bytehammer.com
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