convert pcm stream data to encoded aac data - linux

Question

I tried to convert pulse-audio pcm stream data to aac encoded data using ffmpeg. But after encoding I get noise-full data, not the correct one. Here I post my code, anyone help me with some ideas.
Initial configuration:
av_register_all();
int error;
if ((error = avio_open(&output_io_context,"out.aac",AVIO_FLAG_WRITE))<0) {
printf("could not open output file\n");
}
if (!(output_format_context = avformat_alloc_context())) {
printf("output_format_context error\n");
}
output_format_context->pb = output_io_context;
if(!(output_format_context->oformat = av_guess_format(NULL, "out.aac", NULL))) {
printf("guess format error\n");
}
codec = avcodec_find_encoder(AV_CODEC_ID_AAC);
if (codec == NULL) {
printf("avcodec_find_encoder: ERROR\n");
}
if (!(stream = avformat_new_stream(output_format_context, NULL))) {
printf("stream create error\n");
}
output_codec_context = avcodec_alloc_context3(codec);
if(!output_codec_context) {
printf("output_codec_context is null\n");
}
output_codec_context->channels = CHANNELS;
output_codec_context->channel_layout = av_get_default_channel_layout(CHANNELS);
output_codec_context->sample_rate = SAMPLE_RATE; //input_codec_context->sample_rate;
output_codec_context->sample_fmt = codec->sample_fmts[0];
output_codec_context->bit_rate = 48000; //OUTPUT_BIT_RATE;
stream->time_base.den = SAMPLE_RATE;//input_codec_context->sample_rate;
stream->time_base.num = 1;
if(output_format_context->oformat->flags & AVFMT_GLOBALHEADER)
output_codec_context->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;
if ((error = avcodec_open2(output_codec_context, codec, NULL)) < 0) {
printf("error");
}
error = avcodec_parameters_from_context(stream->codecpar, output_codec_context);
if (write_output_file_header(output_format_context)) {
printf("write header failure...\n");
}
Data encoding:
AVFrame *output_frame;
int frame_pos = 0, ctx_frame_size = output_codec_context->frame_size;
int size = av_samples_get_buffer_size(NULL, CHANNELS,
output_codec_context->frame_size,output_codec_context->sample_fmt, 1);
if((x = avcodec_fill_audio_frame(output_frame, CHANNELS,
output_codec_context->sample_fmt, data, length, 1)) < 0) {
printf("avcodec_fill_audio_frame error : %s\n", av_err2str(x));
}
int data_written;
if (encode_audio_frame(output_frame, output_format_context,
output_codec_context, &data_written)) {
printf("encode_audio_frame error\n");
}
av_frame_free(&output_frame);
helper_function :
int encode_audio_frame(AVFrame *frame,AVFormatContext *output_format_context,
AVCodecContext *output_codec_context, int *data_present)
{
AVPacket output_packet;
int error;
init_packet(&output_packet);
if (frame) {
frame->pts = pts;
pts += frame->nb_samples;
}
error = avcodec_send_frame(output_codec_context, frame);
if (error == AVERROR_EOF) {
error = 0;
goto cleanup;
} else if (error < 0) {
fprintf(stderr, "Could not send packet for encoding (error '%s')\n",
av_err2str(error));
return error;
}
error = avcodec_receive_packet(output_codec_context, &output_packet);
if (error == AVERROR(EAGAIN)) {
error = 0;
goto cleanup;
} else if (error == AVERROR_EOF) {
error = 0;
goto cleanup;
} else if (error < 0) {
fprintf(stderr, "Could not encode frame (error '%s')\n",
av_err2str(error));
goto cleanup;
} else {
*data_present = 1;
}
if (*data_present &&
(error = av_write_frame(output_format_context, &output_packet)) < 0) {
fprintf(stderr, "Could not write frame (error '%s')\n",
av_err2str(error));
goto cleanup;
}
cleanup:
av_packet_unref(&output_packet);
return error;
}
Do we need to fill AVFrame with sizeof(av_samples_get_buffer_size) or context->frame_size ?
TYIA :) !!