No, I have not. Since I've used ffmpeg for some video applications, I was hoping to stick with ffmpeg to keep the dependencies down.
By easier, do you mean that it is necessary to dispense with libavformat and use libavcodec directly in order to read audio files? most of the audio file formats are listed as available formats in ffmpeg (.wav,.mp3, .ogg, .flac), so I figured it would work. When try to use av_free_packet(), it segfaults when it calls the lower level free() functions (double free or heap corruption.) , right when it calls av_destruct_packet(). Can anyone explain how packets are handled with respect to audio files ? and how to free them ? Any pointers or advice would be greatly appreciated. Thanks, dgstarkweather On Mon, 2010-04-05 at 02:21 +0200, XAVIER ANGUERA MIRO wrote: > Just out of curiosity, have you looked into libsndfile? It is a much > easier library when you have to deal with audio-only formats > > X. > > > On 05/04/10 00:11, "david starkweather" <[email protected]> wrote: > > Greetings, > > I am trying to decode an audio stream from audio files (no > video). > > The function is included below. The main idea is to read > nbsecs of audio > from the file's audio stream, converting to sr sample rate and > number > channels. (For now, just one destination channel.) The signal > is put in > the sigbuf float array and returned. (If the array is not big > enough, a > new one is allocated). > > The problem is I cannot free the packets at the end of each > av_read_frame loop without getting double free or corruption > heap > errors. Naturally, it will work if I replace av_free_packet() > with > av_destruct_nofree, but nothing is free'd, leading to obvious > memory > leaks. What is the proper way to free audio packets in this > instance? > I have read the example source code, and everything points to > using > av_free_packet() to release each packet after each > av_read_frame(). > Does anyone recognize any errors here ? > > I am using pure audio files here (wav, mp3, ogg and flac). Is > there > another way to free these packets when decoding audio files? > Or does it > have to be a video container format ? > > Thanks for all your efforts, > D G Starkweather > > code: > > float* readaudio(const char *filename, float *sigbuf, int > &buflen, const > float nbsecs, const int sr, const int channels) > { > av_log_set_level(AV_LOG_QUIET); > av_register_all(); > > AVFormatContext *pFormatCtx; > > // Open file > if(av_open_input_file(&pFormatCtx, filename, NULL, 0, > NULL)!=0){ > buflen=0; > return NULL ; // Couldn't open file > } > > // Retrieve stream information > if(av_find_stream_info(pFormatCtx)<0){ > buflen=0; > av_close_input_file(pFormatCtx); > return NULL; // Couldn't find stream information > } > > //dump_format(pFormatCtx,0,NULL,0);//debugging > function to print > infomation about format > > unsigned int i; > AVCodecContext *pCodecCtx; > > // Find the video stream > int audioStream=-1; > for(i=0; i<pFormatCtx->nb_streams; i++) > { > > if(pFormatCtx->streams[i]->codec->codec_type==CODEC_TYPE_AUDIO) > { > audioStream=i; > break; > } > } > if(audioStream==-1){ > buflen = 0; > av_close_input_file(pFormatCtx); > return NULL; //no video stream > } > > // Get a pointer to the codec context for the audio > stream > pCodecCtx=pFormatCtx->streams[audioStream]->codec; > > int src_sr = pCodecCtx->sample_rate; > int src_channels = pCodecCtx->channels; > int64_t totalduration = > pFormatCtx->streams[audioStream]->duration; > int64_t duration = (nbsecs > 0.0f) ? > (int64_t)(sr*nbsecs) : > totalduration; > duration = (duration <= totalduration) ? duration : > totalduration; > float *buf = NULL; > if (!sigbuf || buflen <= duration){ > buf = (float*)malloc(duration*sizeof(float)); /* > alloc new buffer > */ > buflen = duration; > } else { > buf = sigbuf; /* use buffer handed in param */ > } > AVCodec *pCodec; > > // Find the decoder > pCodec=avcodec_find_decoder(pCodecCtx->codec_id); > if(pCodec==NULL) > { > buflen=0; > av_close_input_file(pFormatCtx); > return NULL ; // Codec not found > } > // Open codec > if(avcodec_open(pCodecCtx, pCodec)<0){ > buflen=0; > av_close_input_file(pFormatCtx); > return NULL; // Could not open codec > } > ReSampleContext *rsctx = > audio_resample_init(channels,src_channels,sr,src_sr); > > uint8_t *inbuf = > (uint8_t*)av_mallocz(AVCODEC_MAX_AUDIO_FRAME_SIZE > > +FF_INPUT_BUFFER_PADDING_SIZE); > uint8_t *outbuf= > (uint8_t*)av_mallocz(AVCODEC_MAX_AUDIO_FRAME_SIZE > > +FF_INPUT_BUFFER_PADDING_SIZE); > int nbsamples, numbytesread, bufsize = > AVCODEC_MAX_AUDIO_FRAME_SIZE; > > int64_t index = 0; > AVPacket packet; > while(av_read_frame(pFormatCtx, &packet)>=0 && index < > duration) > { > while (packet.size > 0){ > nbsamples = bufsize; > numbytesread = > > avcodec_decode_audio2(pCodecCtx,(short*)inbuf,&nbsamples,packet.data,packet.size); > > if (nbsamples > 0){ > int cnt = audio_resample(rsctx, > (short*)outbuf, (short*)inbuf, > nbsamples); > > if (index + cnt > duration){ > duration = index + cnt; > } > for (int i=0;i < cnt;i++){ > short val = ((short*)outbuf)[i]; > buf[index+i] = > (float)(val)/(float)SHRT_MAX; > } > > index += cnt; > } > packet.size -= numbytesread; > packet.data += numbytesread; > } > av_free_packet(&packet); > } > > audio_cleanup: > av_free(inbuf); > audio_resample_close(rsctx); > avcodec_close(pCodecCtx); > av_close_input_file(pFormatCtx); > buflen = (int)index; > > return buf; > } > > > > > > _______________________________________________ > libav-user mailing list > [email protected] > https://lists.mplayerhq.hu/mailman/listinfo/libav-user > _______________________________________________ libav-user mailing list [email protected] https://lists.mplayerhq.hu/mailman/listinfo/libav-user
