Marcelo, have you initialized the ffmpeg using av_register_all()? Also, when building the ffmpeg, have you included encoding/decoding/ parsing support for aac?
Sylvio Em 12/01/2014 16:18, "Marcelo Paes Rech" <[email protected]> escreveu: > Hi guys, > > I am trying to do a stream player in android but I need to read a array of > bytes and then decode every packet of array of bytes. In the end started to > make my AAC decoder work with the ffmpeg example of decoder. But I ran into > the same problem of this guy: > > > http://stackoverflow.com/questions/13499480/decode-aac-to-pcm-with-ffmpeg-on-android > > But the decoder does not work. I am receiving a error as follows: > > TNS filter order %d is greater than maximum %d. > Error while decoding: -1 > > If I use the ffmpeg to decode it works fine. > > ffmpeg -i audio.mp4 audio.wav > > Environment: > ffmpeg 1.2 > Android ndkr9b > > Follows my source code: > > AVCodec *codec; > > AVCodecContext *c= NULL; > > int len; > > FILE *f, *outfile; > > uint8_t inbuf[AUDIO_INBUF_SIZE + FF_INPUT_BUFFER_PADDING_SIZE]; > > AVPacket avpkt; > > AVFrame *decoded_frame = NULL; > > av_log_set_callback(&my_ffmpeg_log); > > av_init_packet(&avpkt); > > > printf("Decode audio file %s to %s\n", filename, outfilename); > > > /* find the mpeg audio decoder */ > > codec = avcodec_find_decoder(AV_CODEC_ID_AAC); > > if (!codec) { > > LOGV("Codec not found\n"); > > return; > > } > > > c = avcodec_alloc_context3(codec); > > c->channels = 2; > > c->sample_rate = 48000; > > > if (!c) { > > LOGV("Could not allocate audio codec context\n"); > > return; > > } > > > /* open it */ > > if (avcodec_open2(c, codec, NULL) < 0) { > > LOGV("Could not open codec\n"); > > return; > > } > > > f = fopen(filename, "rb"); > > if (!f) { > > LOGV("Could not open %s\n", filename); > > return; > > } > > outfile = fopen(outfilename, "wb"); > > if (!outfile) { > > av_free(c); > > return; > > } > > > /* decode until eof */ > > avpkt.data = inbuf; > > avpkt.size = fread(inbuf, 1, AUDIO_INBUF_SIZE, f); > > > while (avpkt.size > 0) { > > int got_frame = 0; > > > if (!decoded_frame) { > > if (!(decoded_frame = avcodec_alloc_frame())) { > > LOGV("Could not allocate audio frame\n"); > > return; > > } > > } else > > avcodec_get_frame_defaults(decoded_frame); > > > len = avcodec_decode_audio4(c, decoded_frame, &got_frame, &avpkt); > > if (len < 0) { > > LOGV("Error while decoding: %d\n", len); > > return; > > } > > if (got_frame) { > > /* if a frame has been decoded, output it */ > > int data_size = av_samples_get_buffer_size(NULL, c->channels, > > > decoded_frame->nb_samples, > > c->sample_fmt, 1); > > fwrite(decoded_frame->data[0], 1, data_size, outfile); > > } > > avpkt.size -= len; > > avpkt.data += len; > > avpkt.dts = > > avpkt.pts = AV_NOPTS_VALUE; > > if (avpkt.size < AUDIO_REFILL_THRESH) { > > /* Refill the input buffer, to avoid trying to decode > > * incomplete frames. Instead of this, one could also use > > * a parser, or use a proper container format through > > * libavformat. */ > > memmove(inbuf, avpkt.data, avpkt.size); > > avpkt.data = inbuf; > > len = fread(avpkt.data + avpkt.size, 1, > > AUDIO_INBUF_SIZE - avpkt.size, f); > > if (len > 0) > > avpkt.size += len; > > } > > } > > > fclose(outfile); > > fclose(f); > > > avcodec_close(c); > > av_free(c); > > avcodec_free_frame(&decoded_frame); > > -- > *Atenciosamente, Marcelo Paes Rech.* > E-mail: [email protected] > Blog: http://marcelopaesrech.blogspot.com > > > _______________________________________________ > Libav-user mailing list > [email protected] > http://ffmpeg.org/mailman/listinfo/libav-user > >
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