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May I ask a question which do not relate to MAD?
by Sherwin Huang 22 Jun '05

22 Jun '05

Dear Rob, Do you have plan to write a fixed-point AAC decoder? Thank you for your answer. Best regards, Sherwin

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seeking in stream
by Vadim Berezniker 21 Jun '05

21 Jun '05

I quickly looked through the sample code and mad.h and I didn't see any facility for seeking. I know I am probably just being blind right now... but how does one seek in a stream?

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how to use it
by dfs sd 16 Jun '05

16 Jun '05

Hi guys! I am new to the MAD and I want to just use it as a decorder,who can help me how I can use it? I have download it on my computer and saw the minimad.c several times but I can't get a idea. There are so many files in the MAD and I am confused by so many files. Thank you for your time. HeWeidong Regards. --------------------------------- DO YOU YAHOO!? 雅虎免费G邮箱－中国最强免费防毒反垃圾超大邮箱

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Without dynamic
by Daniel Levitan 06 Jun '05

06 Jun '05

Hi everyone, After many tries I've finally succeed decoding a mp3 file. But now the sound seems whitout dynamic, I don't know if the problem is the function that I'm using to scale the samples or the routine that I've used to save it to a file (using libsndfile). Can anyone help me ? Thank you ! Daniel #include <stdio.h> #include <stdlib.h> #include <string.h> #include "mad.h" #include "sndfile.h" #define INPUT_BUFFER_SIZE 65535 struct private_data { FILE *file; unsigned char *inputBuffer; int numChannels; int samplerate; int len; float *sample; int icount, ocount; bool firstRun; void *userData; //nao sei pra que vai usar... }; /* The MAD callbacks */ enum mad_flow input_cb (void *_data, struct mad_stream *stream); enum mad_flow output_cb(void *_data, struct mad_header const *header, struct mad_pcm *pcm); enum mad_flow error_cb (void *_data, struct mad_stream *stream, struct mad_frame *frame); inline float scale(mad_fixed_t sample) { return (float) (sample / (float) (1L << MAD_F_FRACBITS)); } int main(){ struct private_data mPrivateData; mad_decoder mDecoder; int chn; const char *filename="saida.wav"; SF_INFO info; SNDFILE *outfile; /* Prepare decoder data, initialize decoder */ mPrivateData.len = 0; mPrivateData.file = fopen("weapon2.mp3","rb"); mPrivateData.inputBuffer = new unsigned char [INPUT_BUFFER_SIZE]; mPrivateData.numChannels = 0; mPrivateData.samplerate = 0; mPrivateData.icount = mPrivateData.ocount = 0; mPrivateData.firstRun = true; mad_decoder_init(&mDecoder, &mPrivateData, input_cb, 0, 0, output_cb, error_cb, 0); if(mad_decoder_run(&mDecoder, MAD_DECODER_MODE_SYNC) == 0) { /* success */ printf("success\n"); //GRAVAR O ARQUIVO COM A SNDFILE info.samplerate = mPrivateData.samplerate; info.channels = mPrivateData.numChannels; //printf("Sample rate %d\n",info.samplerate); //printf("Channels %d\n",info.channels); //info.samplerate = 44100; //info.channels = 1; info.format = 65538; info.sections = 1; info.seekable = 1; info.frames = 0; outfile = sf_open(filename , SFM_WRITE , &info); sf_write_float( outfile, mPrivateData.sample , mPrivateData.len) ; sf_close(outfile); mad_decoder_finish(&mDecoder); /* copy the WaveTrack pointers into the Track pointer list that * we are expected to fill */ //*outTracks = new Track* [mPrivateData.numChannels]; //for(chn = 0; chn < mPrivateData.numChannels; chn++) { // mPrivateData.channels[chn]->Flush(); // (*outTracks)[chn] = mPrivateData.channels[chn]; //} //*outNumTracks = mPrivateData.numChannels; //delete mPrivateData.inputBuffer; //delete[] mPrivateData.channels; //return true; } else { /* failure */ printf("failure\n"); mad_decoder_finish(&mDecoder); /* delete everything */ //for(chn = 0; chn < mPrivateData.numChannels; chn++) // delete mPrivateData.channels[chn]; //delete[] mPrivateData.channels; //delete mPrivateData.inputBuffer; //return false; } //mad_decoder_finish(&mDecoder); /* delete everything */ //for(chn = 0; chn < mPrivateData.numChannels; chn++) //delete mPrivateData.channels[chn]; printf("Input %d Output %d\n",mPrivateData.icount, mPrivateData.ocount); delete mPrivateData.inputBuffer; fclose(mPrivateData.file); return(0); } // // MAD Callbacks // /* The input callback is called when the decoder wants more data. */ enum mad_flow input_cb(void *_data, struct mad_stream *stream) { struct private_data *data = (struct private_data *)_data; data->icount += 1; if(feof(data->file)) return MAD_FLOW_STOP; /* "Each time you refill your buffer, you need to preserve the data in * your existing buffer from stream.next_frame to the end. * * This usually amounts to calling memmove() on this unconsumed portion * of the buffer and appending new data after it, before calling * mad_stream_buffer()" * -- Rob Leslie, on the mad-dev mailing list */ unsigned int unconsumedBytes; if(stream->next_frame) { unconsumedBytes = data->inputBuffer + INPUT_BUFFER_SIZE - stream->next_frame; memmove(data->inputBuffer, stream->next_frame, unconsumedBytes); } else unconsumedBytes = 0; //off_t read = data->file->Read(data->inputBuffer + unconsumedBytes, // IsNPUT_BUFFER_SIZE - unconsumedBytes); //size_t Read(void* buffer, size_t count) int read = fread(data->inputBuffer + unconsumedBytes, 1, INPUT_BUFFER_SIZE - unconsumedBytes, data->file); mad_stream_buffer(stream, data->inputBuffer, read + unconsumedBytes); return MAD_FLOW_CONTINUE; } /* The output callback is called every time the decoder has finished decoding * a frame, allowing us to use the decoded data */ enum mad_flow output_cb(void *_data, struct mad_header const *header, struct mad_pcm *pcm) { int channels, samplerate, samples; struct private_data *data = (struct private_data *)_data; int chn, smpl, cont; float *bufferL, *bufferR; samplerate= pcm->samplerate; channels = pcm->channels; samples = pcm->length; if (data->firstRun){ data->numChannels = channels; data->samplerate = samplerate; data->firstRun = false; } if(data->ocount == 0) data->sample = (float *) malloc(channels*samples*sizeof(float) ); else data->sample = (float *) realloc (data->sample , (data->len + samples)*sizeof(float)*channels ); data->ocount += 1; if (!data->sample){ printf("NAo alocou memo\n"); exit(1); } float *buffer = new float[channels*samples]; if (channels == 1){ for(smpl = 0; smpl < samples; smpl++) buffer[smpl] = scale(pcm->samples[0][smpl]); } if (channels == 2){ bufferL = new float[samples]; bufferR = new float[samples]; for(smpl = 0; smpl < samples; smpl++){ bufferL[smpl] = scale(pcm->samples[0][smpl]); bufferR[smpl] = scale(pcm->samples[1][smpl]); } //Interleaving cont = 0; while (cont < samples){ buffer[cont] = bufferL[cont]; cont++; buffer[cont] = bufferR[cont]; cont++; } } memcpy(data->sample + data->len ,buffer ,channels*samples); data->len +=channels*samples; if (channels == 2){ delete [] bufferL; delete [] bufferR; } delete [] buffer; //Codigo para somente 1 canal /*if(data->ocount == 0) data->sample = (float *) malloc(samples*sizeof(float) ); else data->sample = (float *) realloc (data->sample , (data->len + samples)*sizeof(float) ); data->ocount += 1; if (!data->sample){ printf("NAo alocou memo\n"); exit(1); } float *buffer = new float[samples]; for(smpl = 0; smpl < samples; smpl++) buffer[smpl] = scale(pcm->samples[0][smpl]); memcpy(data->sample + data->len ,buffer ,samples); data->len +=samples;*/ /* If this is the first run, we need to create the WaveTracks that * will hold the data. We do this now because now is the first * moment when we know how many channels there are. */ /* if(!data->channels) { data->channels = new WaveTrack* [channels]; for(chn = 0; chn < channels; chn++) { data->channels[chn] = data->trackFactory->NewWaveTrack(floatSample); data->channels[chn]->SetRate(samplerate); data->channels[chn]->SetChannel(Track::MonoChannel); } /* special case: 2 channels is understood to be stereo if(channels == 2) { data->channels[0]->SetChannel(Track::LeftChannel); data->channels[1]->SetChannel(Track::RightChannel); data->channels[0]->SetLinked(true); } data->numChannels = channels; } else { // This is not the first run, protect us from libmad glitching // on the number of channels channels = data->numChannels; } */ /* TODO: get rid of this by adding fixed-point support to SampleFormat. * For now, we allocate temporary float buffers to convert the fixed * point samples into something we can feed to the WaveTrack. Allocating * big blocks of data like this isn't a great idea, but it's temporary.*/ /* float **channelBuffers = new float* [channels]; for(chn = 0; chn < channels; chn++) channelBuffers[chn] = new float [samples]; for(smpl = 0; smpl < samples; smpl++) for(chn = 0; chn < channels; chn++) channelBuffers[chn][smpl] = scale(pcm->samples[chn][smpl]); for(chn = 0; chn < channels; chn++) data->channels[chn]->Append((samplePtr)channelBuffers[chn], floatSample, samples); for(chn = 0; chn < channels; chn++) delete[] channelBuffers[chn]; delete[] channelBuffers; */ return MAD_FLOW_CONTINUE; } enum mad_flow error_cb(void *_data, struct mad_stream *stream, struct mad_frame *frame) { /* enum mad_flow { MAD_FLOW_CONTINUE = 0x0000, MAD_FLOW_STOP = 0x0010, MAD_FLOW_BREAK = 0x0011, MAD_FLOW_IGNORE = 0x0020 }; */ /* printf("decoding error 0x%04x (%s)\n", stream->error, mad_stream_errorstr(stream)); */ return MAD_FLOW_CONTINUE; /* return MAD_FLOW_BREAK; */ } No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.323 / Virus Database: 267.6.2 - Release Date: 4/6/2005

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MAD Winamp Plugin & ID3v2 Tags
by Daniel Morais 24 May '05

24 May '05

Today, I downloaded the MAD Winamp Plugin to work around major problems I'd been having with Nullsoft's decoder for years. Needless to say, now that everything seems to work, I'm very happy with my sound! However, I noticed that the latest version of the WinAmp plugin, 0.14.2b, doesn't seem to know about MP3 ID3v2 tags (only v1). This is particularly disappointing for me as I've been putting meaningful information in ID3v2s for quite a while now. I can work around this by loading my files with Nullsoft's input plugin (it loads the tags) and never Viewing Tags using the MAD plugin -- unless I do that, it seems to avoid rereading the tags, leaving my original ID3v2s intact in WinAmp. I wondered if there has been any work on this or if any of you had suggestions for getting full ID3v2 support (editing v2 tags would be nice too!). I have some programming experience myself, but I couldn't seem to find the source for the WinAmp plugin itself to look at it. If this is a simple enough change that I could help with it without knowing anything about the existing code base, I'd be more than happy to. I imagine this is a problem that many MAD Winamp plugin users would be frustrated with. Please let me know if anyone has any ideas or suggestions (or is there a newer version somewhere that I'm missing?...I see MAD itself has moved beyond 0.14.2b, but I guess the plugin hasn't been updated...) Thanks very much. - Daniel Morais

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Re: MAD library for TI DSP
by Sherwin Huang 06 May '05

06 May '05

Hi, Grigory, I am trying to port MAD library to ADI-2181, which is 16-bit fixed-point DSP. Would you please send me your implementation for reference? Thank you in advance. Best regards, Sherwin ------------------------------------------------------------------------------------------------------- Hi loboWu! I've sent it to you from several e-mail accounts - but it didn't work - always same - fail to delivery So I try to do same from my yahoo.com e-mail and more http://dsp.neora.ru/cores/mp3.zip Grigory. Thursday, August 5, 2004, 2:28:30 PM, you wrote: l> Hi, Grigory: l> l> l> I am trying to port the MAD library for the TI-54X DSP,too l> And I try to implemnt it with Q1.15, but in vain. l> The output is so terrible with noise. l> l> Could you please send me your implementation for reference ? l> l> I will try my best for it, and put it on DM270 platform! l> l> Thank you very much! l> l> Lobo Wu, ITRI/CCL Taiwan -- Best regards, Grigory mailto:Ryhor@tut.by

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trying to decode a mp3
by a19617＠ualg.pt 04 May '05

04 May '05

hello... I'm working with the libmad library and I'm trying to decode an mp3. I've follow the guidelines in the Install file, the configure and then the make. And I've got no errors. After all that I did: $ make minimad $ ./minimad test.mp3 test.pcm but after that the test.pcm file is empty. Can anybody help me, because I realy don't have a clue why it doesn't work. What am I doing wrong? p.s:I'm using linux, gentoo, to run it. -- Sonia Gonçalves Nº 19617 ESI

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(no subject)
by freeoicq 21 Apr '05

21 Apr '05

mad-dev: Some question about MAD on the 16bit fixed point CPU. Can anyone provide me something about a implemention of MPEG-1 Layer III on it? 　　　　　　　　致礼！　　　　　　　　freeoicq 　　　　　　　　freeoicq(a)huawei.com 　　　　　　　　　　2005-04-21

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MAD on ARM7
by 史志存 21 Mar '05

21 Mar '05

Hi, colleagues, I want to implement a software MP3 decoder. Beacuse MAD gives a fix-pointed implementation, I think I can port it to my ARM7 platform. Are there someone who are working on it? Would u please give me the performance profile? btw, beside MAD, is there other open source fix-pointed decoder? Thanks for your enthusiasm. Jason

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Arm cross compile
by Andrew Douglas 13 Mar '05

13 Mar '05

Hi I wonder if anyone can help. I'm trying to cross compile madplay-0.15.2b for an arm9 processor running a 2.4.18 kernel. I'm building the source on a redhat9 host. I do the following: $ ./configure && make && make install for libmad and libid3tag. Then I try configuring madplay with the following command: $ RANLIB=arm-linux-ranlib CC=arm-linux-gcc ./configure --host=arm-linux and get the following error: checking mad.h usability... no checking mad.h presence... no checking for mad.h... no configure: error: mad.h was not found *** You must first install libmad before you can build this package. I include the libmad dir with CFLAG and then it complains about libid3. I include this then it complains: configure: error: libmad was not found This doesn't seem right. Is there something I've done wrong. Any hints would be appreciated. TIA Andy Douglas

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