mad-dev

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682 discussions

Re: [mad-dev] madplay: How to select a specific ALSA card
by joker1999 19 Sep '08

19 Sep '08

Thomas Reitmayr-2 wrote: > > Hi, > I just started using madplay a short while ago and I was wondering > whether it is possible to select an ALSA card different from the default > one for audio output. >>From looking at the code it seems like the PATH part of --output is > used, however > madplay --output=alsa:plughw:1 myfile.mp3 > chokes as the TYPE alsa does not exist (in audio.c). > > Did I miss something in the command line or is this a bug / missing > feature (for which I could provide a patch)? > > Thanks, > -Thomas > > > You can try: madplay -o wave:- something.mp3 | aplay Regards -- View this message in context: http://www.nabble.com/madplay%3A-How-to-select-a-specific-ALSA-card-tp14695… Sent from the Mad - Dev mailing list archive at Nabble.com.

1 1

Converting fixed point numbers
by Reiner 12 Jul '08

12 Jul '08

Hello, madplay uses the following in audio_linear_round /* quantize and scale */ return sample >> (MAD_F_FRACBITS + 1 - bits); if bits is 16 (unsigned short e,g) we will shift 13 bits, whats about the 3 MSB bits ??? With this I think only the first digit of the whole part is taken, what is with the sign and the other 2 digits from the whole part, are they never used? Thanks Reiner

2 1

libmad documentation project
by Michelle Baert 09 Jun '08

09 Jun '08

Hi, I recently started working on a small audio processing project. I chose to use mad library for dealing with mp3 files. Since I needed to study it for getting enough understanding of it, and the package TODO file calls for contributors to documentation project, I tried to make my notes useful for the programmers community. One thing I like in java programming is the documentation facility, which generates browseable API documentation directly from comments inserted in source files. The same functionality is available for C/C++ and other languages, with Doxygen (open-source) program, so I decided to use it. You'll find all details and files here: http://m.baert.free.fr/contrib/docs/libmad/ Your comments and suggestions are welcome. TIA for them. -- Best regards, Michelle Baert

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madplayer cross-compile on ARM11 compilation error "impossible constraint in 'asm'"
by Jianfeng Li 21 May '08

21 May '08

Hi, All, I am now trying to cross-compile the madplayer on an arm11 based embedded system. The compiler I am using is from montavista mobilinux 5.0. When I tried to build the madplay, for the file resample.c, I got 2 errors on line 89 and 104. The error information is "impossible constraint in 'asm'. When I tried to build the libmad, I also met the same error. But it seems that during compilation if "FPM-ARM" option is enabled then the error is solved. But here I checked the Makefile, there is no such option of "FPM" as in the Makefile of libmad. So can anybody help to tell me why this error happens, it is due to the compiler's constraint/requirement? How should I try to solve this error? Is "trying to enable the FPM" a good option to try? Many thanks in advance! Regards, Jeffen _________________________________________________________________ Invite your mail contacts to join your friends list with Windows Live Spaces. It's easy! http://spaces.live.com/spacesapi.aspx?wx_action=create&wx_url=/friends.aspx…

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Stream Decode
by Alexey V Paramonov 20 May '08

20 May '08

Hi! I have the following question - is it possible to decode a stream using MAD? For example: I have shoutcast stream and I read it in small portions (say, 40960 bytes), decode it and encode back to mp3 using LAME (chaning bitrate, samplerate, etc...). The problem is that I have to call mad_stream_buffer with each piece of stream data, decode/encode it and then if I listen the resulting mp3 file - I notice "noise" between theese small decoded pieces. Seems like it happen not only with VBR files. Pseudo-code: while ( bytes = read(file_id, buff, SHOUT_BUFFER_LENGTH ) ) { mad_stream_buffer(&stream, buff, bytes ); while(1){ mad_frame_decode(&frame,&stream) .... int ret = lame_encode_buffer( lameGlobalFlags, left, right, nsamples, mp3Buf, mp3Size ); // Play resulting mp3 data } } Is it possible to decode a stream without theese "edge effects" ? May be I should not re-initialize MAD buffer after each read in some way? Please help.

1 0

Initite loop bug in libid3tag-0.15.0b
by Kentaro Oda 30 Jan '08

30 Jan '08

Hi, for the maintainer libid3tag. I found an infinite loop bug in libid3tag-0.15.0b library, which causes memory overflow. The problem occurs when parsing an ID3_FIELD_TYPE_STRINGLIST field, specifically when data to be parsed is ended with '\0'. In this case, **ptr == 0, but the condition end - *ptr is 1 so loop continues infinitely. *** field.c 2003-04-19 09:14:33.000000000 +0900 --- field-patched.c 2008-01-13 16:08:22.000000000 +0900 *************** *** 291,297 **** end = *ptr + length; ! while (end - *ptr > 0) { ucs4 = id3_parse_string(ptr, end - *ptr, *encoding, 0); if (ucs4 == 0) goto fail; --- 291,297 ---- end = *ptr + length; ! while (end - *ptr > 0 && **ptr != '\0') { ucs4 = id3_parse_string(ptr, end - *ptr, *encoding, 0); if (ucs4 == 0) goto fail; -- Kentaro Oda

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madplay: How to select a specific ALSA card
by Thomas Reitmayr 08 Jan '08

08 Jan '08

Hi, I just started using madplay a short while ago and I was wondering whether it is possible to select an ALSA card different from the default one for audio output. >From looking at the code it seems like the PATH part of --output is used, however madplay --output=alsa:plughw:1 myfile.mp3 chokes as the TYPE alsa does not exist (in audio.c). Did I miss something in the command line or is this a bug / missing feature (for which I could provide a patch)? Thanks, -Thomas

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memory leaking in mad_layer_III?
by Niels Roosen 27 Sep '07

27 Sep '07

Hello, In the past while I have been rounding up development for a streaming server for the radio station I am part of. However, during the last stress testing phase I noticed that memory usage slowly increases as streaming clients disconnect and connect. At first I was convinced it was some leak in my own code but valgrind didn't confirm that this was happening. However, I noticed that there was quite some lost memory originating in the mad_layer_III() function in layer3.c: ==5957== 7,701 bytes in 3 blocks are definitely lost in loss record 68 of 85 ==5957== at 0x481B72C: malloc (in /usr/lib/valgrind/x86-linux/vgpreload_memcheck.so) ==5957== by 0x485F1C1: mad_layer_III (in /usr/lib/libmad.so.0.2.1) ==5957== by 0x485429A: mad_frame_decode (in /usr/lib/libmad.so.0.2.1) ==5957== by 0x7775: decode_data (server.c:318) ==5957== by 0x9387: server (server.c:980) ==5957== by 0x4874164: pthread_start_thread (in /lib/libpthread-0.10.so) ==5957== by 0x4A176E9: clone (in /lib/libc-2.3.6.so) ==5957== 13,824 bytes in 3 blocks are definitely lost in loss record 71 of 85 ==5957== at 0x481CE9E: calloc (in /usr/lib/valgrind/x86-linux/vgpreload_memcheck.so) ==5957== by 0x485F215: mad_layer_III (in /usr/lib/libmad.so.0.2.1) ==5957== by 0x485429A: mad_frame_decode (in /usr/lib/libmad.so.0.2.1) ==5957== by 0x7775: decode_data (server.c:318) ==5957== by 0x9387: server (server.c:980) ==5957== by 0x4874164: pthread_start_thread (in /lib/libpthread-0.10.so) ==5957== by 0x4A176E9: clone (in /lib/libc-2.3.6.so) One would assume this is just caused by memory not being free()'d up when the program exits, but the longer the server has run, the more memory was definitely lost, indicating that the memory loss originates here. I use the mad library in such a way that I manually line up data and decode this data per frame, which is required to ensure certain functionality of the streaming server. I made sure that when I reinitialize the decoder every time I run the mad_*_finish() operations on all the structs I use before running mad_*_init() again. However, I suspect that not all allocated memory is freed as it should. Am I missing something or could it be that using mad this way hasn't been tested excessively enough and is it true that the memory alllocated in mad_layer_III() on lines 2530 and 2538 (libmad version 0.15.1b) is not freed properly? I would be delighted to either get confirmation on this or some advice on how to solve this issue. If more information is required I will do my best to provide this. Regards, Niels Roosen -- "One of the most frightening things about your true nerd, for many people, is not that he's socially inept - because everybody's been there - but rather his complete lack of embarrassment about it." - Randy Waterhouse in Cryptonomicon n: Arnie / Arnicus | e: niels(a)okkernoot.net w: http://okkernoot.net/ | irc: irc.okkernoot.net #geek | icq: 31817478 aim: arniea a arniea | msn: arnie(a)okkernoot.net http://jungletrain.net/ - 24/7 Drum and Bass Internet Radio

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Calculation error
by Tim Dylla 07 Sep '07

07 Sep '07

Hi List, I did some testing on my own implementation of mp3, and used MAD as reference. Therefore I generated a sine-sweep from 0Hz to 22kHz ( as far as this is possible ;) ) at 0dBFS, and 48kHz Samplerate. Then I encoded it with Lame for a single Channel MPEG-1 Layer III Bitstream, 48kHz Samplerate, 320kbps Bitrate. (I don't know the exact Lame-version and optionstring anymore, but I didn't use anything weird) As expected, the resulting bitstream can be decoded without any obvious error with my Decoder and also with the dist10 decoder. MAD in contrast introduces a perceivable error right on the first half wave of the sine sweep. I think, it's an overflow in the requantization or the IMDCT (when I remember right, the huffman decoder produces very high output at this moment). I used MAD 0.15.2b in conjunction with MADFrontend on a 32bit-Windoze-machine. The option string was -b24 -d I'm not a friend of posting large files on a mailing list without any agreement with the moderator, so if someone is interested in the bitstream and the resulting raw file, pleas feel free to send me a message. With regards, Tim Dylla

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libMad Vs ffmpeg
by Chulwoong Jeon 24 Aug '07

24 Aug '07

I am porting MP3 at ARM9 platform. If I use only fixed-point(Integer), I wonder which one is faster between MAD mp3 decoding and ffmpeg mp3 decoding. CW

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Re: [mad-dev] Calculation error
by Tim Dylla 24 Aug '07

24 Aug '07

Andre schrieb: > Tim Dylla <tim.dylla(a)gmx.net> wrote: >> MAD in contrast introduces a perceivable error right on the first >> half wave of the sine sweep. I think, it's an overflow in the >> requantization or the IMDCT (when I remember right, the huffman >> decoder produces very high output at this moment). > > Just a hunch, but if you are able to build MAD from source, maybe you > could try the follow patch posted to the list a few years back... > perhaps it helps ?? > > diff -ruN mad-0.14.2b_orig/libmad/layer3.c > mad-0.14.2b/libmad/layer3.c > --- mad-0.14.2b_orig/libmad/layer3.c Thu Nov 8 15:28:02 2001 > +++ mad-0.14.2b/libmad/layer3.c Tue Mar 25 15:21:12 2003 > @@ -910,7 +910,7 @@ > fprintf(stderr, "requantize overflow (%f * 2^%d)\n", > mad_f_todouble(requantized), exp); > # endif > - requantized = MAD_F_MAX; > + requantized = MAD_F_MAX / 2; > } > else > requantized <<= exp; Nope, same Result. I'm sorry, I'm still sticking with my diploma, so I don't have the time for extensive tests right now. Maybe someone can confirm the problem and have a look at? Feel free to contact me for the Bitstream in Question and the correct decoder output. Best, Tim Dylla > > > Andre > -- > >

1 1

[Offtopic] Stories about Process Change
by Christopher Oezbek 23 Aug '07

23 Aug '07

Hi MAD-Hackers, Hi Rob, I am researching what happens if a developer attempts to change the way a project works (in contrast to changing the software being developed) and have picked MAD as a small OSS project in long-term existence that seems to use a closed development strategy. I would be very interested in hearing if there was ever the need to modify the development process and if yes how successful have those changes been. I put up a survey of 13 questions (should take between 10 and 20 minutes) at http://survey.mi.fu-berlin.de/public/survey.php?name=changeEpisodeSurvey but you can also email me directly if you prefer. Thank you very much, Christopher Oezbek P.S.: If you are thinking about making such a change or are currently on it, feel free to drop me a note so I can share what I have learned so far. P.P.S.: Results will be made available at http://www.inf.fu-berlin.de/w/SE/SurveyChange over the next month, where you can also find further information. -- Christopher Oezbek Freie Universität Berlin http://www.inf.fu-berlin.de/~oezbek/

1 0

problem with the seeking over
by cab 17 Aug '07

17 Aug '07

Hi, Ok, I managed to seek in a frame Thanks for your help Cab

1 0

Re: [mad-dev] mad-dev Digest, seeking problem with libmad
by cab 16 Aug '07

16 Aug '07

Hi, You said I had to map the current file at a seek point but I just can't do itn there's a problem with the function mmap itself here's my source: (from minimad) int main(int argc, char *argv[]) { struct stat stat; void *fdm; if (argc != 1) return 1; int fd; FILE*fichier=fopen("essai.mp3","r+"); fd=fileno(fichier); lseek(fd,10,SEEK_SET); if (fstat(fd, &stat) == -1 || stat.st_size == 0) return 2; fdm = mmap(0,stat.st_size/2-1 , PROT_READ, MAP_SHARED, fd,stat.st_size/2); if (fdm == MAP_FAILED) return 3; decode(fdm, stat.st_size/2-1); if (munmap(fdm, stat.st_size/2-1) == -1) return 4; return 0; } I've got an error return 22 I know it's because the offset is not set to 0 thanks for your help

1 0

test-bitstreams
by Tim Dylla 30 Jul '07

30 Jul '07

Hi list, since I'm doing final tests on my mp3-decoder implementation for a Blackfin DSP, I'm searching for teststreams coming with a decoded reference. The original ISO/IEC/Fraunhofer-teststream package contains only the decoded reference of the compliance bitstream. But (as you certainly know) it doesn't test the whole functionality of MP3 (not even short blocks), and the decoder in this package is crap (limited compliance, far away from being a reference). I'm searching for more bitstreams in company with their decoded reference to avoid adding up deviance of the used Encoder to my test results. Has anyone some resource regarding this? I'm interested in MPEG1 and 2 - Layer III bitstreams. Thanks in advance, Tim

2 1

Computation Speed
by Tim Dylla 23 Jul '07

23 Jul '07

Hello List, I'm new to this list, let me introduce myself. I'm working on my Diploma for Applied Informatics, I have to implement the MP3-Decoder for the Analog Devices BF-533 DSP. It's the the final testphase regarding accuracy, where MAD really convinces. Since I want to leave the needed MIPS of my Algorithm low, I had to make some compromise between Accuracy and Speed. Now I wonder: how much MIPS needs MAD? Say, for the x86-Version optimized for Accuracy? Has anyone measured this? Or is there a Webpage with all that info, and I was just dumb? :) Best Regards, Tim Dylla

2 2

compiling libmad in c++
by julien ricard 05 Jul '07

05 Jul '07

hi, how can I compile libmad using a c++ compiler? using g++ and -D__cplusplus gives errors. For example: $ g++ -c -D__cplusplus -DFPM_INTEL decoder.c decoder.c: In function 'void mad_decoder_init(mad_decoder*, void*, mad_flow (*)(void*, mad_stream*), mad_flow (*)(void*, const mad_header*), mad_flow (*)(void*, const mad_stream*, mad_frame*), mad_flow (*)(void*, const mad_header*, mad_pcm*), mad_flow (*)(void*, mad_stream*, mad_frame*), mad_flow (*)(void*, void*, unsigned int*))': decoder.c:76: error: invalid conversion from 'int' to 'mad_decoder_mode' decoder.c: In function 'mad_flow error_default(void*, mad_stream*, mad_frame*)': decoder.c:299: error: invalid conversion from 'void*' to 'int*' decoder.c: In function 'int mad_decoder_run(mad_decoder*, mad_decoder_mode)': decoder.c:553: error: invalid conversion from 'void*' to 'mad_decoder::<anonymous struct>*' thanks jul

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mad_stream_skip to seek to a new frame
by www.bigcab＠free.fr 19 Jun '07

19 Jun '07

I think I can use mad_stream_skip to seek a new frame but I always got a lost sync error I don't understand?

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libmad know the current bytes read or decoded
by www.bigcab＠free.fr 18 Jun '07

18 Jun '07

here I would like to know if it could be possible to know if we could retrive the current byte read (an equivalent of ftell) aren't there any functions for it? thanks very much

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problem with the seeking
by www.bigcab＠free.fr 18 Jun '07

18 Jun '07

Hi here I've got a VBR (variable bitrate file ) so I can't seek the right frame It's not as if I had a Pcm file can someone help me?

2 1

libid3tag
by cab 16 Jun '07

16 Jun '07

Hello, Ok I'm here to report a few bugs with libid3tag The function id3_file_open works half the time file=id3_file_open(input,ID3_FILE_MODE_READONLY); if (file==NULL) { printf("file error(id3_tag_size fun)\n"); return 1; } this line often return 1 and prints file error each time I execute this command, I check the file path and it's correct so I concluded that the bug was coming from this function. Thanks. Best Regards Cab

1 0

seeking error
by www.bigcab＠free.fr 08 Jun '07

08 Jun '07

rob wrote : Sadeesh Kumar wrote: > In libmad, when the song is playing, how can i jump to different > differnt location and start to play from that location?. I find it, there is > no direct easy interface to do this. To jump to a new location using the low-level API: 1. Seek to an approximate location in the file and load your buffer with data from the new position. 2. Call mad_stream_buffer() to update the stream pointers for the modified buffer. 3. Call mad_frame_mute() and mad_synth_mute(). At this point you can simply resume decoding. However, to avoid any unpleasant burps in the audio, unless you have seeked to the very beginning of the file, it is best to: 4. Call mad_frame_decode() successfully at least twice. Note that the first few calls will probably yield recoverable decoding errors which you can safely ignore. 5. Call mad_synth_frame() once and discard the resulting PCM. The MAD plug-in for Winamp does all of this if you're looking for an example. Cheers, -rob but the problem is I've got these errors decoding error 0x0101 (lost synchronization) at byte offset 4096 decoding error 0x0235 (bad main_data_begin pointer) at byte offset 4203 and then my player doesn't work any more, there's a bug what is wrong ? here my source code int seek(state*status /*structure to store the data*/,int value) { printf("seeking in mp3 started\n"); struct mad_stream *stream; stream=status->stream; struct mad_frame frame; frame=status->mad_frame; struct mad_synth synth; synth=status->synth; printf("%lf is the value in percent\n",value); pthread_mutex_lock(&mutex); int seek_bit; int bitrate=status->bitrate; int samplerate=status->srate; int frame_size=144*bitrate/samplerate; seek_bit=percent_to_bits(status->mp3_file_size,(int)value); int reste=seek_bit-status->cur_bit; int factor=reste/frame_size; seek_bit=status->cur_bit+factor*frame_size+4*factor; printf("mad stream buffer call\n"); mad_stream_buffer(stream,status->start_mp3+seek_bit,status->mp3_file_size-seek_bit); status->cur_bit=seek_bit; mad_frame_mute(&frame); mad_synth_mute(&synth); int result,i=0; result=mad_frame_decode(&frame,stream); while((result!=0)||(i<2)) { result=mad_frame_decode(&frame,stream); i+=1; } mad_synth_frame(&synth,&frame); pthread_mutex_unlock(&mutex); printf("seeking in mp3 ended\n"); return 0; } what is wrong with my source code?

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MAD_ERROR_BADHUFFDATA before MAD_ERROR_BUFLEN
by Radek Bartoň 22 May '07

22 May '07

Hello. I tried to incorporate Madlld low-level API demonstration code into my MP3 player, but I'm experiencing a weird problem. When data stored by mad_stream_buffer to MAD are going to run out MAD_ERROR_BADHUFFDATA is generated right before MAD_ERROR_BUFLEN. Then when new data are sent to MAD MAD_ERROR_BADCRC and MAD_ERROR_BADDATAPTR occur. Since then no other errors occur until next data are consumed. This produces slight skip in otherwise fluent playback. I checked twice that my code which I'm attaching is equivalent to Madlld's one. Please, if you'll notice a reasong of this behaviour in my code, let me know since I'm little desperate about this. Thank you. -- Bc. Radek Bartoň Faculty of Information Technology Brno University of Technology E-mail: xbarto33(a)stud.fit.vutbr.cz Web: http://blackhex.no-ip.org Jabber: blackhex(a)jabber.cz

2 6

Question on New IMDCT Algorithm
by 岷兰王 09 Apr '07

09 Apr '07

Hi, I have been reading the new IMDCT algorithm in libmad-0.15.1b. I can understand how it works. But what I can't get how the "scale[]" array is produced. It seems not to be the same with other arrays in layer3.c, such as cs[], ca[], etc. Can any body help me on this question?? Thank you! --------------------------------- Mp3疯狂搜-新歌热歌高速下

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Re: mad-dev Digest, Vol 20, Issue 2
by Li xiaochao.Jeff 31 Mar '07

31 Mar '07

For algorithm, I think it is almost the best. But for implementation, there might be another optimization. Remember, the choice of algorithm is much dependent upon the hardware architecture. Regards, Li xiaochao.Jeff ----- Original Message ----- From: <mad-dev-request(a)lists.mars.org> To: <mad-dev(a)lists.mars.org> Sent: Friday, March 30, 2007 3:00 AM Subject: mad-dev Digest, Vol 20, Issue 2 > 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. Synthesis filter bank in libmad (Karthika) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Thu, 29 Mar 2007 07:45:58 +0100 (BST) > From: Karthika <karthika_lakshmi(a)yahoo.co.in> > Subject: [mad-dev] Synthesis filter bank in libmad > To: MAD <mad-dev(a)lists.mars.org> > Message-ID: <333191.28381.qm(a)web7906.mail.in.yahoo.com> > Content-Type: text/plain; charset=iso-8859-1 > > Helloo, > I am trying to understand the synthesis filter > bank in the LIBMAD. I could not understand in what > order the D coefficients are used to obtain the PCM > samples. Also,half of these coefficients have some > relation to the other half. Then, why all the > coefficients are being used in D.dat file? What is the > significance of even and odd pointers? How is the > windowing operation between U vector and D vector > taking place? > > Also, I would like to know if we can further > optimize the LIBMAD source code. For e.g, by using > Cooley-Tukey algorithm for the calculation of > fastdct() and dct32() . Could anyone suggest any > further optimization techniques for porting LIBMAD on > an ARM processor? > > Any help will be greatly appreciated. > > Thanks and Regards, > Karthika Lakshmi. > > > > > > __________________________________________________________ > Yahoo! India Answers: Share what you know. Learn something new > http://in.answers.yahoo.com/ > > > End of mad-dev Digest, Vol 20, Issue 2 > ************************************** >

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