Question

I am trying to make http streaming program.
So I follow this code at this.
However, when i decode, only one frame is decoded.
I think I need call back function.
Do you know how to make a call back function?
I know ‘asf’ packet’s call back function is like int read_data(void *opaque, char *buf, int buf_size)

But the other formats(mp3, ogg, aac, ..) doesn’t work..

Please help me.

Any advice or comment are very appreciated.

#include <stdio.h>
#include <stdlib.h>
#include <libavcodec/avcodec.h>
#include <libavformat/avformat.h>
#include <libavdevice/avdevice.h>

int main(int argc, char **argv)
{
        static AVInputFormat *file_iformat;
        static AVFormatContext *pFormatCtx;
        AVFormatParameters params;

        AVCodecContext *pCodecCtx;
        AVCodec *pCodec;

        const char url[] = "http://listen.radionomy.com/feelingfloyd";

        avcodec_register_all();
        avdevice_register_all();
        av_register_all();

        av_log_set_level(AV_LOG_VERBOSE);

        file_iformat = av_find_input_format("mp3"); /* mp3 demuxer */
        if (!file_iformat)
        {
                fprintf(stderr, "Unknown input format: %s\n", &url[0]);
                exit(1);
        }

        //file_iformat->flags |= AVFMT_NOFILE; /* ??? */
        params.prealloced_context = 0;

        if (av_open_input_file(&pFormatCtx, &url[0], file_iformat, 0, &params) != 0)
        {
                fprintf(stderr, "err 1\n");
                exit(2);
        }

        /* poulates AVFormatContex structure */
        if (av_find_stream_info(pFormatCtx) < 0)
        {
                fprintf(stderr, "err 2\n");
        }

        /* sanity check (1 stream) */
        if (pFormatCtx->nb_streams != 1 &&
                        pFormatCtx->streams[0]->codec->codec_type != AVMEDIA_TYPE_AUDIO)
        {
                fprintf(stderr, "err 3\n");
        }

        pCodecCtx = pFormatCtx->streams[0]->codec;

        /* find decoder for input audio stream */
        pCodec = avcodec_find_decoder(pCodecCtx->codec_id);
        if (pCodec == NULL)
        {
                fprintf(stderr, "err 4: unsupported codec\n");
        }

        if (pCodec->capabilities & CODEC_CAP_TRUNCATED)
                pCodecCtx->flags |= CODEC_FLAG_TRUNCATED;

        if (avcodec_open(pCodecCtx, pCodec) < 0)
        {
                fprintf(stderr, "err 5\n");
        }

        {
                uint8_t *pAudioBuffer;
                AVPacket pkt;

                int ret;
                int data_size = 2 * AVCODEC_MAX_AUDIO_FRAME_SIZE;

                av_init_packet(&pkt);
                //pkt.data=NULL;
                //pkt.size=0;
                //pkt.stream_index = 0;

                pAudioBuffer = av_malloc(data_size * sizeof(int16_t));

                while (av_read_frame(pFormatCtx, &pkt) == 0) {
                        //data_size = AVCODEC_MAX_AUDIO_FRAME_SIZE;
                        ret = avcodec_decode_audio3(pFormatCtx->streams[pkt.stream_index]->codec,
                                        (int16_t *)pAudioBuffer, &data_size, &pkt);

                        /* got an error (-32) here */
                        if (ret < 0) {
                                av_strerror(ret, (char *)pAudioBuffer, data_size);
                                fprintf(stderr, "err 6 (%s)\n", pAudioBuffer);
                                break;
                        }

                        printf("size=%d, stream_index=%d |ret=%d data_size=%d\n",
                                        pkt.size, pkt.stream_index, ret, data_size);
                        av_free_packet(&pkt);
                }

                av_free(pAudioBuffer);
        }

        avcodec_close(pCodecCtx);
        av_close_input_file(pFormatCtx);

        return 0;
}

Answer 1

I figure out this problem by using av_open_input_file.
I got a this problem when I made a iphone app that play http audio streaming. And the above code didn’t work. only played just some of audio frame. it also means there are so many buffering.
However After using iphone audio callback function and large audio buffer, it works fine.
Those who are curious about the final code send me a messege.