mad-dev January 2006

mad-dev@lists.mars.org

7 participants
9 discussions

error linking to libid3tag
by Alex 27 Jan '06

27 Jan '06

Hello, I have compiled libid3tag with mingw on windows. When linking with libid3tag, I have this error : configure:20404: gcc -o conftest.exe -g -O2 conftest.c -lid3tag -lmad -lvorbisfile >&5 c:/mp3splt_mingw/bin/../lib/gcc-lib/mingw32/3.2.3/../../../libid3tag.a(util.o)(.text+0x152): In function `id3_util_compress': C:/MinGW/home/admin/mp3splt/libid3tag/util.c:106: undefined reference to `compress2' c:/mp3splt_mingw/bin/../lib/gcc-lib/mingw32/3.2.3/../../../libid3tag.a(util.o)(.text+0x1e4): In function `id3_util_decompress': C:/MinGW/home/admin/mp3splt/libid3tag/util.c:139: undefined reference to `uncompress' You have probably already encountered this error , but I have searched on the net, have found a solution but did not understand it. If I link with -lz it works but why do I have to link with -lz (libid3tag should already be linked with -lz) ? Any help would be appreciate. -- Munteanu Alexandru --------------------------------- Nouveau : téléphonez moins cher avec Yahoo! Messenger ! Découvez les tarifs exceptionnels pour appeler la France et l'international.Téléchargez la version beta.

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[mad-dev] Re: seeking, jumping to a certain position within mp3 file
by Li Xiaochao.Jeff 27 Jan '06

27 Jan '06

As I know, the mp3 file is organized in frame base which means it might not jump to a specific sample number(if it is the original input, compressed data). And the decoding process is much dependent upon the integrity of the frame. By the way, I am also wondering what is the scenario it might be useful about this feature? Best regards, Li xiaochao.Jeff ----- Original Message ----- From: <mad-dev-request(a)lists.mars.org> To: <mad-dev(a)lists.mars.org> Sent: Thursday, January 19, 2006 4:00 AM Subject: mad-dev Digest, Vol 8, Issue 4 > Send mad-dev mailing list submissions to > mad-dev(a)lists.mars.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://www.mars.org/bin/mailman/listinfo/mad-dev > or, via email, send a message with subject or body 'help' to > mad-dev-request(a)lists.mars.org > > You can reach the person managing the list at > mad-dev-owner(a)lists.mars.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of mad-dev digest..." > > > Today's Topics: > > 1. seeking, jumping to a certain position within mp3 file > (felber.michael(a)gmx.net) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Wed, 18 Jan 2006 20:39:37 +0100 > From: <felber.michael(a)gmx.net> > Subject: [mad-dev] seeking, jumping to a certain position within mp3 > file > To: <mad-dev(a)lists.mars.org> > Message-ID: <200601181934.k0IJYNEe008160(a)surveyor.mars.org> > Content-Type: text/plain; charset="us-ascii" > > Hi, > > does libmad provide the possibility to jump to a certain position within a > mp3-file? > > Is it possible to jump to a certain sample number (e.g. 44100, 88200) within > a mp3-file? > > Ciao, Michael > > > > > End of mad-dev Digest, Vol 8, Issue 4 > *************************************

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Newbie - How to skip backward in steam?
by Jim Witte 27 Jan '06

27 Jan '06

Hi, I'm working on the Apple Newton port of MAD. How do you get the player to skip backward in the stream? I can skip forward using mad_stream_skip, but it won't take negative offsets. I tried calling mad_stream_buffer again to restart the current buffer playing from the beginning, but this causes the sound to stop (this may be because of the way the Newton sound architecture operates - if MAD is unable to decode anything, the buffer back to the NOS is empty, and it shuts down the sound codec). Can I just set stream->this_frame and stream- >next_frame directly? What does stream_sync do? Thanks, Jim Witte jim3416(a)hotpop.com

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can I use MAD to fade and split an MP3?
by Chris Dawson 26 Jan '06

26 Jan '06

I am interested in using MAD to fade out at an arbitrary point in an arbitrary MP3 file and then potentially split the file at the end of the fade out. Is MAD appropriate for this? If not, does anyone have suggestions for other OSS libraries that can assist with this kind of thing. I could probably use sox after ripping the file to wav with mplayer, but the roundtrip conversion from MP3 -> WAV -> MP3 would probably degrade the file significantly, so I would prefer not to do that. Thanks, Chris

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check if a file is a valid mp3
by Munteanu Alexandru 24 Jan '06

24 Jan '06

Hello, is there any way to check if a file is a mp3 file using libmad ? -- Munteanu Alexandru Ionut <io_alex_2002 @no_spam@ yahoo.fr>

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seeking, jumping to a certain position within mp3 file
by felber.michael＠gmx.net 18 Jan '06

18 Jan '06

Hi, does libmad provide the possibility to jump to a certain position within a mp3-file? Is it possible to jump to a certain sample number (e.g. 44100, 88200) within a mp3-file? Ciao, Michael

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The detail information of synth_full in libmad
by Li Xiaochao.Jeff 17 Jan '06

17 Jan '06

Hello, Sir I am quite interest in the detail info of synth_full in libmad. I got the general ideal of how it funtions, such as dct32 for matrix operation and good arrangement of data in order to reduce the multiplication for windows. However, I would like to get more detail information so that I could do further optimization to this module in ARM platform. And seek the possiblities for using WMMX or other technique. Best regards, Li xiaochao.Jeff

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Using MAD with small buffers
by Daniel Kraft 14 Jan '06

14 Jan '06

Hi! I want to use MAD for audio playback, and looked at the minimad.c example. My problem is that I can't map the whole input to memory at once (as is done there) but I have to use fixed size blocks (size of my input buffer), which I feed to MAD and refill in my input callback. I've adapted the minimad.c code (my version is included below) to use fixed size blocks of input; if that buffer size is greater then my file size it works quite fine; but when it is smaller, I get loads of errors (mostly lost synchronization) after a few blocks are read. As there's not much documentation about that, I hope you can help me: When I use stream_buffer(...), how long is that buffer data needed to be available? Can I clear (or reuse) that buffer the next time my input callback is called? Thank you very much for your help! Yours, Daniel /* ************************************************************* */ # include <stdio.h> # include <unistd.h> # include <sys/stat.h> # include <sys/mman.h> #include <assert.h> #include <malloc.h> # include "mad.h" #define BUFFER_SIZE (1<<10) typedef struct { unsigned char* start; unsigned long length; FILE* in; } mydata; static int decode(mydata*); int main(int argc, char *argv[]) { mydata buffer; buffer.in=fopen(argv[1], "rb"); assert(buffer.in); buffer.start=malloc(BUFFER_SIZE); buffer.length=0; decode(&buffer); free(buffer.start); fclose(buffer.in); return 0; } static enum mad_flow input(void *data, struct mad_stream *stream) { mydata *buffer = data; if(!buffer->length) { buffer->length=fread(buffer->start, sizeof(*buffer->start), BUFFER_SIZE, buffer->in); if(!buffer->length) return MAD_FLOW_STOP; } mad_stream_buffer(stream, buffer->start, buffer->length); buffer->length=0; return MAD_FLOW_CONTINUE; } static inline signed int scale(mad_fixed_t sample) { /* round */ sample += (1L << (MAD_F_FRACBITS - 16)); /* clip */ if (sample >= MAD_F_ONE) sample = MAD_F_ONE - 1; else if (sample < -MAD_F_ONE) sample = -MAD_F_ONE; /* quantize */ return sample >> (MAD_F_FRACBITS + 1 - 16); } static enum mad_flow output(void *data, struct mad_header const *header, struct mad_pcm *pcm) { unsigned int nchannels, nsamples; mad_fixed_t const *left_ch, *right_ch; /* pcm->samplerate contains the sampling frequency */ nchannels = pcm->channels; nsamples = pcm->length; left_ch = pcm->samples[0]; right_ch = pcm->samples[1]; while (nsamples--) { signed int sample; /* output sample(s) in 16-bit signed little-endian PCM */ sample = scale(*left_ch++); putchar((sample >> 0) & 0xff); putchar((sample >> 8) & 0xff); if (nchannels == 2) { sample = scale(*right_ch++); putchar((sample >> 0) & 0xff); putchar((sample >> 8) & 0xff); } } return MAD_FLOW_CONTINUE; } static enum mad_flow error(void *data, struct mad_stream *stream, struct mad_frame *frame) { mydata *buffer = data; fprintf(stderr, "decoding error 0x%04x (%s) at byte offset %u\n", stream->error, mad_stream_errorstr(stream), stream->this_frame - buffer->start); /* return MAD_FLOW_BREAK here to stop decoding (and propagate an error) */ return MAD_FLOW_CONTINUE; } static int decode(mydata* buffer) { struct mad_decoder decoder; int result; /* configure input, output, and error functions */ mad_decoder_init(&decoder, buffer, input, 0 /* header */, 0 /* filter */, output, error, 0 /* message */); /* start decoding */ result = mad_decoder_run(&decoder, MAD_DECODER_MODE_SYNC); /* release the decoder */ mad_decoder_finish(&decoder); return result; }

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About subband synthesis
by g924352 05 Jan '06

05 Jan '06

Hello all, I am trying to improve the performance of the MP3 decoder. One idea is to improve the subband synthesis. In the MAD, the subroutine, synth_full(), process the subband synthesis. In this subroutine, it can be found that the data (dct32 output) is accessed sequentially. However, the coefficients (synthesis window) is not accessed sequential. I am trying to modify the subroutine in order to access the coefficients sequentially. My question is that where can i find the document about the flow of the subroutine or suggestion. thanks a lot!! Best Regards

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mad-dev January 2006