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
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