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682 discussions

On additions to libid3tag
by Linus Walleij 05 Sep '04

05 Sep '04

I've been having all kinds of fun with libid3tag lately. However there were two things I couldn't do: 1. Add tags to files that do not already have ID3 tags. 2. Remove tags from files. This seem to stem from the fact that it's a concieved as a modification library only, not a creation or destruction library... So I wrote external code for adding and removing tags. Now I thought that this could perhaps be included in libid3tag. Would this be good? If you like the idea, I could adopt my code for libid3tag (follow your code standards etc) but just in case you think it's worth it. Below are copies of the routines, which are written for the glib portability library (will have to be changed of course) for inspection. static gboolean addid3tag(gchar *path) { // File descriptors... gint f1, f2, f3; gboolean retval = FALSE; gchar template[128]; // g_print("Adding a new ID3 tag to %s\n", path); f1 = (gint) open(path, O_RDONLY, 0); if (f1 < 0) return FALSE; // g_print("Opened file f1...\n"); // template = g_build_filename(g_get_tmp_dir(), "gnomadXXXXXX"); strcpy(template, g_get_tmp_dir()); template[strlen(template)+1] = '\0'; template[strlen(template)] = G_DIR_SEPARATOR; strcat(template, "gnomadXXXXXX"); // Temporary file f2 = g_mkstemp(template); // g_print("Opened temporary file f2...\n"); if (f2 >= 0) { register gchar *buffer; gchar blankv2tag[0x0c0a]; guint bufsiz; // A blank ID3v2 tag with title set to "x" gchar v2head[] = {0x49, 0x44, 0x33, 0x03, 0x00, 0x00, 0x00, 0x00, 0x0c, 0x00, // size 0x0100 0x54, 0x50, 0x45, 0x31, 0x00, 0x00, 0x00, 0x0a, 0x00, 0x00, 0x00, 0x78}; gchar blankv1tag[] = {'T', 'A', 'G', 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00 }; // Allocate a copying buffer for (bufsiz = 0x8000; bufsiz >= 128; bufsiz >>= 1) { buffer = (gchar *) g_malloc(bufsiz); if (buffer) break; } // g_print("Allocated a buffer of %d bytes...\n", bufsiz); // Add a dummy tag // g_print("Adding a dummy ID3v2 tag...\n"); memset(blankv2tag, 0, 0x0c0a); memcpy(blankv2tag, v2head, 22); if (0x0c0a == write(f2,blankv2tag,0x0c0a)) { // Copy the entire file //g_print("Copying the original file...\n"); while (1) { register guint n; n = read(f1,buffer,bufsiz); if (n == -1) break; if (n == 0) { retval = TRUE; break; } if (n != write(f2,buffer,n)) break; } } // g_print("Adding a dummy ID3v1 tag...\n"); write(f2,blankv1tag,128); if (retval) { close(f1); // Rewind the temporary file lseek(f2, 0, SEEK_SET); // Then copy the file back again // g_print("Copying the file back...\n"); f3 = (gint) creat(path, (mode_t) CREATE_FILEMODE); if (f3) { // g_print("Creat() on original file succeeded...\n"); while (1) { register guint n; n = read(f2,buffer,bufsiz); if (n == -1) break; if (n == 0) { retval = TRUE; break; } if (n != write(f3,buffer,n)) break; } close(f3); } else { // g_print("Error copying back!\n"); retval = FALSE; } close(f2); } else { close(f1); close(f2); } // g_print("Deleting %s\n", template); unlink(template); // Free the buffer g_free(buffer); return retval; } close(f1); return FALSE; } void remove_tag_from_mp3file(gchar *path) { // File descriptors... gint f1, f2, f3; gchar template[128]; guint header_taglength = 0; guint footer_taglength = 0; guint filelength = 0; register gchar *buffer; guint bufsiz; gboolean retval = FALSE; // not used yet // template = g_build_filename(g_get_tmp_dir(), "gnomadXXXXXX"); //FEL strcpy(template, g_get_tmp_dir()); template[strlen(template)+1] = '\0'; template[strlen(template)] = G_DIR_SEPARATOR; strcat(template, "gnomadXXXXXX"); // g_print("Removing ID3 tags from %s\n", path); f1 = (gint) open(path, O_RDONLY, 0); if (f1 < 0) { return; } // g_print("Opened file f1...\n"); // Allocate a copying buffer for (bufsiz = 0x8000; bufsiz >= 128; bufsiz >>= 1) { buffer = (gchar *) g_malloc(bufsiz); if (buffer) break; } // Temporary file f2 = g_mkstemp(template); // g_print("Opened temporary file f2...\n"); if (f2 >= 0) { guchar tag[10]; gint n; // g_print("Allocated a buffer of %d bytes...\n", bufsiz); // g_print("Looking for ID3v2 header tag\n"); n = read(f1,tag,10); if (n == 10 && tag[0] == 'I' && tag[1] == 'D' && tag[2] == '3') { // g_print("Found ID3v2 tag header...\n"); // Get tag length from the tag header_taglength = (tag[6] << 24) + (tag[7] << 16) + (tag[8] << 8) + tag[9]; // This needs some heuristics... first scan until only zeroes are found. // while () n = read(f1,buffer,bufsiz); if (n > 4) { guint m = 0; gboolean foundzeros = FALSE; gboolean founddata = FALSE; // Look for 8 consecutive zeroes while (!foundzeros && (m < n)) { if (buffer[m] == 0x00 && buffer[m+1] == 0x00 && buffer[m+2] == 0x00 && buffer[m+3] == 0x00 && buffer[m+4] == 0x00 && buffer[m+5] == 0x00 && buffer[m+6] == 0x00 && buffer[m+7] == 0x00) { foundzeros = TRUE; } m += 8; } // g_print("Found zeros\n"); // Then look for first non-zero while (!founddata && (m < n)) { if (buffer[m] == '3' && buffer[m+1] == 'D' && buffer[m+2] == 'I') { // We found a footer, wind past it and say that we found data. m += 10; founddata = TRUE; } else if (buffer[m] != 0x00) { founddata = TRUE; // Tag is now m bytes long! } else { m++; } } // If length was properly detected, we now know the length. if (founddata) { // g_print("Found data\n"); if (header_taglength != m) { g_print("Bad header tag! Given length: %d, detected length: %d\n", header_taglength, m); } // Adjust header taglength header_taglength = m; // Wind past any RIFF header too header_taglength += riff_header_size(&buffer[m], n-m); } // At last, include the header too. header_taglength += 10; // g_print("ID3v2 header (and any RIFF header) %d bytes\n", header_taglength); } } else { // Any plain RIFF header is removed too. gint n; n = read(f1,buffer,bufsiz); header_taglength = riff_header_size(&buffer[0], n); } // As we move to the end of the file, detect the filelength filelength = lseek(f1, 0, SEEK_END); if (filelength > 0) { // Then detect the length of any ID3v1 tag // g_print("Detecting ID3v1 tag\n"); if (lseek(f1, -128, SEEK_END) > 0) { guchar tag[3]; gint n; n = read(f1,tag,3); if (n == 3 && tag[0] == 'T' && tag[1] == 'A' && tag[2] == 'G') { // g_print("Found ID3v1 tag footer...\n"); footer_taglength = 128; } } // Then detect the length of any ID3v2 footer tag // g_print("Detecting ID3v2 footer tag\n"); if (lseek(f1, -10-footer_taglength, SEEK_END) > 0) { guchar tag[10]; guint footlen; gint n; n = read(f1,tag,10); if (n == 10 && tag[0] == '3' && tag[1] == 'D' && tag[2] == 'I') { // g_print("Found ID3v2 footer tag\n"); footlen = (tag[6] << 24) + (tag[7] << 16) + (tag[8] << 8) + tag[9]; // First remove the tag headers footer_taglength += 20; // Then remove the indicated length (no looking for bad tag here) footer_taglength += footlen; } } // g_print("Header %d bytes, footer %d bytes to be removed.\n", header_taglength, footer_taglength); // Next skip past the header, and copy until we reach the footer if (lseek(f1, header_taglength, SEEK_SET) > 0) { guint remain = filelength - header_taglength - footer_taglength; while (remain) { register guint n; if (remain > bufsiz) { n = read(f1,buffer,bufsiz); } else { n = read(f1,buffer,remain); } if (n == -1) break; if (n == 0) { retval = TRUE; break; } if (n != write(f2,buffer,n)) break; remain -= n; } } } close(f1); // Then copy back the stripped file // Rewind the temporary file lseek(f2, 0, SEEK_SET); // g_print("Copying the file back...\n"); f3 = (gint) creat(path, (mode_t) CREATE_FILEMODE); if (f3) { // g_print("Creat() on original file succeeded...\n"); while (1) { register guint n; n = read(f2,buffer,bufsiz); if (n == -1) break; if (n == 0) { retval = TRUE; break; } if (n != write(f3,buffer,n)) break; } close(f3); } else { g_print("Error copying file back!\n"); retval = FALSE; } close(f2); // g_print("Deleting %s\n", template); unlink(template); } else { // In case we couldn't open f2 retval = FALSE; close(f1); } g_free(buffer); }

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Using MAD for MP3 Streaming decoding
by balaji shanmugam 01 Sep '04

01 Sep '04

I am using MAD library for my application which basically decodes the MP3 streams. I am using HELIX server (from RealNetworks) as the MP3 streaming server. After I started the streaming (the helix server streaming a local MP3 file encoded at Constant Bit rate of 128Kbps and 441.KHz sampling freq). The streaming is done using RTSP/RTP protocol. I captured the streamed out packets and found out that the length of each RTP payload is varying. The first few stream packets are containing less than 417/418 bytes of Mp3 audio data. I have taken care of copying the unprocessed/left out MP3 data of previous frame into the new frame but I am always getting the LOST_SYNC error. Will someone help me out in processing the MP3 data that is streamed out. Regards Balaji Shanmugam _________________________________________________________________ Drive away out of trouble. http://citibank.mediaturf.net/carcash2/landing/key_landing.jsp?referrer=HMt… Get a loan against your existing car.

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question about left channel frequency information in stereo mode
by Shiming Dong 16 Aug '04

16 Aug '04

Hi there: I met with a new problem in the stereo mode: The two channels have different sound information and I only need to detect a tone with the specific frequency in the left channel. What changes shall I make for the following formula if I have two channels and each channel has 576 units pcm information. Thank you! shiming > 1. I use mad library to get the pcm data of a marker tone, each frame > contain 576 samples. I turned each 16 bits integer samples into floating > point by divided by 32768. And I put it into a floating point array > pcm[576]. > > 2. I tried to do fourier transformation to those pcm data: > Fk = sum (pcm[i] * cos(2*pi*k*i / N)) > > N = 576 > i: from 0 to N > k: from 0 to N / 2 > F[288]: is a floating point array > Thank you! shiming --------------------------------- Do you Yahoo!? New and Improved Yahoo! Mail - 100MB free storage!

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[patch] add pkg-config support
by Eivind Magnus Hvidevold 07 Aug '04

07 Aug '04

Here is a patch for adding pkg-config support to libmad. It would make life a bit easier for distro maintainers if this was applied. In case you didn't know, pkg-config is a tool for providing LDFLAGS and CFLAGS for packages using shared libraries. It's on freedesktop.org. Debian has already been distributing the pkg-config file mad.pc with libmad for some time, and people developing on debian (notably xmms2 developers) have started relying on this support being present, causing some confusion for people installing from source and on some BSDs which do not provide mad.pc (google: pkgconfig libmad). EMH

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mp3 bitstream error
by Grigory A. 06 Aug '04

06 Aug '04

Hi! I am trying to play attached bitstream using mimimad application. and minimad say to me decoding error 0x0235 at byte offset 0 MAD_ERROR_BADDATAPTR = 0x0235, /* bad main_data_begin pointer */ but my winamp play it well. What is the problem? I attach only firs 16 kB of bitstream. in start of file there is a header of 0x240 bytes I "remove" it by simple seeking the file pointer. The problem is located in function int mad_layer_III(MAD_STREAM *stream, MAD_FRAME *frame) but where exactly here error = III_sideinfo(&stream->ptr, nch, header->flags & MAD_FLAG_LSF_EXT, &si, &data_bitlen, &priv_bitlen); or here - // -------- find main_data of this frame ----------- frame_space = stream->next_frame - mad_bit_nextbyte(&stream->ptr); ... I don't know. And one of possibilities is - wrong bitstream? but winamp play it well. So could some one help me understand what is the problem? -- Best regards, Grigory mailto:Ryhor@tut.by

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Re: MAD library for TI DSP
by Grigory A. 05 Aug '04

05 Aug '04

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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Re: [mad-dev] Re: mad-dev digest, Vol 1 #368 - 2 msgs
by armin.gerritsen＠philips.com 29 Jul '04

29 Jul '04

> is that the DAC runs autonomously, and interrupts when it wants more > data. > I can't 'choose' how fast to give it samples. As the CPU and DAC > clocks > won't be exactly synchronised, over time I'll end up with either too > many > or too few samples in my buffer. Not if you buffer in between. Since decoding MPEG-frames is independant of clocking, you can run this as fast or slow as you like. Just decode a bit faster if the buffer you're using to feed the DAC is running below a certain treshhold and halt decoding of input-frames if the buffer gets to full. Or what I did, decode always as fast as possible and try to keep the input-buffers as full as possible. This will allow usage of smaller buffers. The only problem there is, that of course you need to be in control of your MPEG-stream. That is, it needs to be from an asynchronous source like a disk and not from a synchronous like a live network-stream (like shoutcast). But in that case, the problem is not the difference between the CPU clock and the DAC clock, but the source-clock (for instance shoutcast-server) and your DAC-clock. That is a standard, but not easy to solve problem. If that is the case, indeed you have this problem: > I guess I need to add or delete samples, either by changing the > buffer or > by adjusting the resampling ratio slightly. My question is when to > do it > and how much to adjust. You may not believe how many commercial products just use one buffer and when it runs out, stall the whole system untill new data is there, and simply trow samples away. Usually the clock-differences are so small we are talking about just a few samples per hour or less! Of course if your product doesn't allow this, you need to take care of this in a better way, but the solution is hard. Look for instance how it is done in network protocols like RTP/RTCP/etc. Regards, Armin

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RE: [mad-dev] Re: mad-dev digest, Vol 1 #368 - 2 msgs
by Nigel Sinfield 29 Jul '04

29 Jul '04

> > I'm planning to use MAD in a streaming application. I have no > > control over > > the incoming data rate, although it is ostensibly CBR data with a > > fixed > > sample rate. Similarly, the processing rate of the decoded > samples is > > controlled by the DAC. So inevitably sooner or later I'll get a > > buffer > > overflow or underflow. > > So if I understand, the input can be for instance 128kbps or > 160kbps, but > the DAC expects a fixed rate of PCM samples? I don't see a > problem then, > since the rate of PCM samples is independent on the bitrate of the > MPEG-stream. So your PCM-samples will be delivered to the DAC > with always > the same speed. Thanks. I do know the sample rate (in fact, I need to resample the data from typically 48000Hz mono to 22050Hz stereo after the decode). My problem is that the DAC runs autonomously, and interrupts when it wants more data. I can't 'choose' how fast to give it samples. As the CPU and DAC clocks won't be exactly synchronised, over time I'll end up with either too many or too few samples in my buffer. I guess I need to add or delete samples, either by changing the buffer or by adjusting the resampling ratio slightly. My question is when to do it and how much to adjust. The contents of this email and any attachments are sent for the personal attention of the addressee(s) only and may be confidential. If you are not the intended addressee, any use, disclosure or copying of this email and any attachments is unauthorised - please notify the sender by return and delete the message. Any representations or commitments expressed in this email are subject to contract. ntl Group Limited

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More efficient way of random seeking in an mp3 file?
by Robert John Shepherd 29 Jul '04

29 Jul '04

Currently in my MP3 player (using libmad) I have to scan through an entire mp3 to calculate an accurate playtime and build up a random access seek table. On the platform I'm using, this can be very slow taking 3 minutes plus for a 35 minute vbr file. Is there a better way of calculating the length of an mp3 if it isn't in the id3 tag, and/or a better way of jumping to a certain time frame within an mp3 without having scanned the entire thing beforehand? TIA! Yours Unwhettedly, Robert John Shepherd. Editor DVD REVIEWER The UK's BIGGEST Online DVD Magazine http://www.dvd.reviewer.co.uk For a copy of my Public PGP key, email: pgp(a)robertsworld.org.uk

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Re: mad-dev digest, Vol 1 #368 - 2 msgs
by armin.gerritsen＠philips.com 29 Jul '04

29 Jul '04

> I'm planning to use MAD in a streaming application. I have no > control over > the incoming data rate, although it is ostensibly CBR data with a > fixed > sample rate. Similarly, the processing rate of the decoded samples is > controlled by the DAC. So inevitably sooner or later I'll get a > buffer > overflow or underflow. So if I understand, the input can be for instance 128kbps or 160kbps, but the DAC expects a fixed rate of PCM samples? I don't see a problem then, since the rate of PCM samples is independent on the bitrate of the MPEG-stream. So your PCM-samples will be delivered to the DAC with always the same speed. The problem can occur, if the data has also an unknown samplerate (32kHz or 44,1kHz for instance). I had this problem with a network-streaming-device. The MPEG input could and would vary from time to time and this would happen completely randomly. As long as you have some buffering in front of MAD, the decoder has no problem in handling this since it will decode CBR frame by frame. Handling sample-rate changes is more difficult. I added also a small amount of double buffering after the MAD library and I added a small check after the MAD decoder, to see whether the samplerate changed compared to the previous sample. (And for instance 'stereoness' - although that is easy to handle: simply copy the channel to L+R). If a samplerate change was detected, it would switch buffers at once to have the decoded frames with the new samplerate go into the second buffer, while the DAC was still fed from the first. When the first buffer was emptied under a certain level, I would activate a samplerate conversion function on the second buffer. When the first buffer was then empty, the DAC would be fed from that moment on with the converted second buffer, and new decoded samples would enter the first buffer and the process would repeat itself until that buffer was full or another sample-rate conversion was detected. Regards, Armin

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TI 55X Implementation
by balaji shanmugam 29 Jul '04

29 Jul '04

Hi Could anyone send me across information about porting the MADLIB for 16 bit fixed point DSPs (either the 54x or 55x) Regards Balaji Shanmugam _________________________________________________________________ NRIs ! http://go.msnserver.com/IN/52616.asp FREE money transfers to India.

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Streaming rate matching
by Nigel Sinfield 28 Jul '04

28 Jul '04

Hi, I'm planning to use MAD in a streaming application. I have no control over the incoming data rate, although it is ostensibly CBR data with a fixed sample rate. Similarly, the processing rate of the decoded samples is controlled by the DAC. So inevitably sooner or later I'll get a buffer overflow or underflow. Any suggestions how best to handle this? I'm sure someone must have been here before and I don't want to reinvent wheels. Thanks. The contents of this email and any attachments are sent for the personal attention of the addressee(s) only and may be confidential. If you are not the intended addressee, any use, disclosure or copying of this email and any attachments is unauthorised - please notify the sender by return and delete the message. Any representations or commitments expressed in this email are subject to contract. ntl Group Limited

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Number format for sbsamples from IMDCT (Q4.28 to 16 bit PCM samples)
by balaji shanmugam 27 Jul '04

27 Jul '04

Hi Rob I am new in using the MAD Library and upon working with the code, I have got doubt in the Scaling (1st operand shifted right by 12 and 2nd operand shifted right by 16) being done during the mad_f_mul operation. I like to know about the sbsamples from the IMDCT routine. Is it in Q4.28 format. If so why we need to scale it down during SYNTHESIS Filter operation. I have read the comments in fixed.h telling that " * The scale factors are constant as this is not used with SSO. * * Pre-rounding is required to stay within the limits of compliance. " I like to know how this scaling has been taken care in the further calculations. Please find below my doubt. I have defined as FPM_DEFAULT and doesnt define OPT_SPEED.[OPT_SSO will be defined as FPM_DEFAULT is defined]. With that the mad_f_mul (A,B) is defined as [((A+(1L<<11))>>12) * ((B+(1L<<15))>>16)] This has been used during the synth_full operation. When the synth_full function calls the dct32() function, sbsamples<2><2><576> (in Q28.4 format) from IMDCT is passed on to the function where it will be multiplied with costab<1..31>. During this multiplication the mad_f_mul MACRO is used. After that the result from dct32 is stored as the filter output. While storing the filter output, the multiplied value is scaled down by a factor of 12. Then windowing[D array values being scaled down by 12] was applied over the filter ouput and the PCM output is stored with a scale down of 2. The operation of SSO is understandable but the intial scaling that happens with the mad_f_mul MACRO needs explanation. Thanks in advance Balaji Shanmugam _________________________________________________________________ News, views and events. Hot downloads n pictures. http://www.msn.co.in/Cinema/ Get it all at MSN Cinema!

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Re: [mad-dev] MAD library for TI-54X DSPs
by balaji shanmugam 21 Jul '04

21 Jul '04

Hi I too have started doing most of the stuffs in asm rather than going for C functions. The first thing that I am concentrating is the SYNTH function. Bt still since i am following the LONG calculation(Q28.4) instead of Q15.1, mine is still vulnerable for timing issues. It will be gr8 help if you could send me across your version (16 bit arithmetic and Q15.1 for 55X) of MAD decoder for my reference. Thanks in advance Balaji Shanmugam >From: "Grigory A." <Ryhor(a)tut.by> >Reply-To: "Grigory A." <Ryhor(a)tut.by> >To: "balaji shanmugam" <balajiami(a)hotmail.com> >CC: mad-dev(a)lists.mars.org >Subject: Re: [mad-dev] MAD library for TI-54X DSPs >Date: Wed, 21 Jul 2004 09:58:50 +0900 > >Hi Balaji! > >I have ported MAD to TI-55xx. I used Q1.15 format and 16 bits >arithmetics for all calculation. As result of this is: > >limited accuracy ISO/IEC 11172-3 audio decoder > >according to >http://www.underbit.com/resources/mpeg/audio/compliance/ > >Yes - synthesis step of decoding is most calculation intensive part. >I've used a little different method to port MAD to TI-55xx DSP - >instead of providing "arithmetics macro" for given platform I've >rewrite on DSP asm all calculation intensive functions (8 or 9 ?). So >I think this is a better way in case if someone need close to optimal >implementation for given hardware platform. Only you need to do is >to rewrite these asm functions for your platform. > >I can send you my implementation of decoder. And maybe it is better to put >it to >MAD site? Because I've sent it several times before to other people >and this question is arisen again and again :). In additional I have >same style implementation of MAD for sp3R5m DSP (www.3dsp.com) this >one is 32 DSP core and in this case "fully compliant" decoder is >obtained. > > > >Tuesday, July 20, 2004, 11:25:28 PM, you wrote: > >bs> Hi >bs> I am trying to port the MAD library for the TI-54X DSP and running into >bs> tight timing issues. When analysed, I found out that the 54X DSP takes >more >bs> instructions to perform LONG calculations at the SYNTHESIS FILTER Bank >stage >bs> as it is a 16 bit processor [but with a 40 bit accumulator]. As it is >bs> required to perform 72 times the DCT32 and the remaining MATRIX >calculations >bs> at the SYNTH stage, I like to optimize the synthesis filter bank. >bs> I am not sure about using the Q28.4 format being used by MAD. Is there >bs> anyway we can use the Q15.1 format for the calculations? >bs> I like to know is there any hints or clues for the LONG calculations >that I >bs> can make use of during my 54X porting phase. Also any thought about >using >bs> the NORM instruction to round the final samples to 16 bit will be >helpful. > >bs> Regards >bs> Balaji Shanmugam > >bs> _________________________________________________________________ >bs> Marriage? >bs> http://www.bharatmatrimony.com/cgi-bin/bmclicks1.cgi?74 Join >bs> BharatMatrimony.com for free. > > > >-- >Best regards, > Grigory mailto:Ryhor@tut.by > _________________________________________________________________ Get head-hunted by 10,000 recruiters. http://go.msnserver.com/IN/46246.asp Post your CV on naukri.com today.

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MAD library for TI-54X DSPs
by balaji shanmugam 21 Jul '04

21 Jul '04

Hi I am trying to port the MAD library for the TI-54X DSP and running into tight timing issues. When analysed, I found out that the 54X DSP takes more instructions to perform LONG calculations at the SYNTHESIS FILTER Bank stage as it is a 16 bit processor [but with a 40 bit accumulator]. As it is required to perform 72 times the DCT32 and the remaining MATRIX calculations at the SYNTH stage, I like to optimize the synthesis filter bank. I am not sure about using the Q28.4 format being used by MAD. Is there anyway we can use the Q15.1 format for the calculations? I like to know is there any hints or clues for the LONG calculations that I can make use of during my 54X porting phase. Also any thought about using the NORM instruction to round the final samples to 16 bit will be helpful. Regards Balaji Shanmugam _________________________________________________________________ Marriage? http://www.bharatmatrimony.com/cgi-bin/bmclicks1.cgi?74 Join BharatMatrimony.com for free.

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Qns abt MAD Library support for ARM environment
by balaji shanmugam 20 Jul '04

20 Jul '04

Hi I have recently started using the MAD decoder. My requirement is to use the MP3 library for the ARM 32 bit processor environement. I have compiled the library with the option FPM_ARM selected (I have modified the MACRO functions for my compiler support keeping the assembly instructions the same). But the audio output was not clear at all. Then I had selected FPM_DEFAULT. With that I am able to hear a feeble (but with noise) audio. /**** mad_f_mul ****/ #define FPM_ARM # define SIZEOF_INT 4 # define SIZEOF_LONG 4 # define SIZEOF_LONG_LONG 8 # define MAD_F_FRACBITS 28 #define MAD_F_SCALEBITS MAD_F_FRACBITS static mad_fixed_t mad_f_mul(mad_fixed_t x,mad_fixed_t y) { mad_fixed_t result; mad_fixed64hi_t hi; mad_fixed64lo_t lo; _asm(-1000," smull %0,%1,%3,%4 movs %0,lsr(%0,%5) adc %2,%0,lsl(%1,%6) ", __reg_gen(__obj_modify(lo)), __reg_gen(__obj_modify(hi)), __reg_gen(__obj_assign(result)), __reg_gen(x), __reg_gen(y), MAD_F_SCALEBITS, (32 - MAD_F_SCALEBITS) ); return result; } /****Ends****/ I like to know if I am missing something? Thanks Balaji Shanmugam _________________________________________________________________ Reach out to millions of buyers. http://go.msnserver.com/IN/52047.asp With zero investment.

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determining the first scale factor band with short windows
by Rafael Ávila de Espíndola 13 Jul '04

13 Jul '04

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Rob, I noticed that libmad count up to 36 in III_reorder to determine the first scale factor band that should be reordered in a mixed block. The attached patch precomputes this. I don't expect it to improve performance in any measurable way but (IMHO) it makes the code a bit easier to read. Thanks for libmad, Rafael Ávila de Espíndola -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.2.4 (GNU/Linux) iD8DBQFA8zvTLlrfGJ8JUHwRAlmMAJwKzXgUoXPUM+U8u2l4cByyHPjRYgCfWuEZ Jw7Qs4lVqbC9wNHixrDaqFs= =Pu6p -----END PGP SIGNATURE-----

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16bit MAD
by Erik Gustavsson 02 Jul '04

02 Jul '04

Would it be possible to convert MAD into a (limited compliance) decoder using mostly 16bit fixed-point math? Specifically I'd like to port MAD to a ADI Blackfin DSP, which has 32bit registers and ALU but only 16x16 multipliers (with 40bit accumulators).

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RE: [mad-dev] A little help needed on code optimisation to deal with delay slots
by Joolz [RSD] 02 Jul '04

02 Jul '04

I would like to have a look please Gregory. I have already started the port // MLO( x, y); ld %r0, [x] ld %r1, [y] mul64 %r0, %r1 ld %r2, [%mhi] // MLA( x1, y1); ld %r0, [x1] ld %r1, [y1] mul64 %r0, %r1 add %r2, [%mhi] // MLA( x2, y2); ld %r0, [x2] ld %r1, [y2] mul64 %r0, %r1 add %r2, [%mhi] Into ld %r0, [x] ld %r1, [y] mul64 %r0, %r1 ld %r0, [x1] ld %r1, [y1] ld %r2, [%mhi] mul64 %r0, %r1 ld %r0, [x2] ld %r1, [y2] add %r2, [%mhi] mul64 %r0, %r1 Saves 2 cycles per multiply, only another 8 to go Best Regards Julian Gardner RSD Communications Ltd ============================ Please do not send me any html messages as these may be deleted at our mail server without notice to sender or recipient. Please ONLY send me file attachments with any messages with extensions .zip, .rar, or .pdf as any other file types may be deleted at our mail server without notice to sender or recipient. > -----Original Message----- > From: Grigory A. [mailto:Ryhor@tut.by] > Sent: Friday, July 02, 2004 2:34 AM > To: Joolz [RSD] > Cc: mad-dev(a)lists.mars.org > Subject: Re: [mad-dev] A little help needed on code > optimisation to deal with delay slots > > Hi Joolz! > > You have BIG problem :) 10 cycles stall every mul. > The most "multiplication consuming" functions are > - IMDCT - actually it isn't too much > - synthesis filter > > last one has a lot of multiplication > all other relatively free from mult operation. May be stereo > processing has a little. > > If you have short of calculation power I can recommend you to > do same as I did. I've separated calculation functions and > implement them on asm. > I can send the source code of of MAD with such improvements > that I used for > - tms320vc55xx and > - sp3R5 3DSP cores. > > > -- > Best regards, > Grigory mailto:Ryhor@tut.by > >

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A little help needed on code optimisation to deal with delay slots
by Joolz [RSD] 02 Jul '04

02 Jul '04

I have ported the library to our hardware platform ( a set-top box with ide interface and ARC8 processor) but I have found that we are running short of horsepower. The problem stems from the fact that when I use the multiplier it stalls the chip for 10cycles on every multiply mul64 %r0, %r1 lsl %r0, [%mhi], 4 --> Stalls until multiplier finishes I have started looking at reordering the code as I can use other instructions in these 10 cycles <from synth.c dct32:282> // t69 = t33 + t34; t89 = MUL(t33 - t34, costab4); mad_f_mul_j( t33 - t34, costab4); // We have 10 cycles to wait t69 = t33 + t34; // 2-3 t70 = t35 - t36; // 2-3 t89 = mad_f_mul_r(); // t70 = t35 + t36; t90 = MUL(t35 - t36, costab28); // t70 = t35 - t36; mad_f_mul_j( t70, costab28); // We have 10 cycles to wait t70 = t35 + t36; // 2-3 t113 = t69 + t70; // 2-3 t71 = t37 - t38; // 2-3 t90 = mad_f_mul_r(); This gives me savings of around 6 cycles per multiply, so we only lose 4 cycles. What I would like know is which routine/s should I convert first to save the maximum amount of time Joolz

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question about decode mp3-encoded wav
by shiming dong 22 Jun '04

22 Jun '04

Hi there: I tried to get the raw pcm data from a mp3-encoded riff wav file. Can I use the libmad to do that? I mean, except to the header of the file, is each frame header and data the same as that in the mp3 file? And where can I get such file's format description? Thank you! shiming _________________________________________________________________ FREE pop-up blocking with the new MSN Toolbar get it now! http://toolbar.msn.click-url.com/go/onm00200415ave/direct/01/

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Re: Re: [mad-dev] Compiling minimad.c under Windows
by Some One 06 Jun '04

06 Jun '04

Thanks alot. This works perfectly even though it does give a warning saying that "FDM_INTEL" is redefined in "mad.h". However, it decodes an mp3 to pcm fine using "./minimad <track.mp3> track.pcm". >From: Rob Leslie <rob(a)mars.org> >To: mad-dev(a)lists.mars.org >Subject: Re: [mad-dev] Compiling minimad.c under Windows >Date: Sun, 6 Jun 2004 12:45:29 -0700 > >On Jun 6, 2004, at 2:34 AM, Some One wrote: >>Hi, I'm trying to compile minimad.c using Cygwin and GCC. >> >>I've copied the "libmad-0.51.1b" folder into the Cygwin root as "mad" and >>followed all the steps in the "INSTALL" file (configure, make, make >>install). It does all that fine (makes the Makefile and config file, >>creates the objects in the "./deps" folder, installs the mad library in >>"usr/local/lib" and puts "mad.h" in "usr/local/include"). >> >>However, when a try "make minimad" in the "mad" folder, it returns: >>"gcc -Wall -march=i486 -g -O -fforce-mem -fforce-addr -fthread-jumps >>-fcse-follow -jumps -fcse-skip-blocks -fexpensive-optimizations -fregmove >>-fschedule-insns2 -fstrength-reduce >>minimad.c -o minimad >>/cygdrive/c/DOCUME~1/USERNAME/LOCALS~1/Temp/ccyf7LL5.o(.text+0xc3): In >>function `input': >>/mad/minimad.c:92: undefined reference to `_mad_stream_buffer' >>/cygdrive/c/DOCUME~1/USERNAME/LOCALS~1/Temp/ccyf7LL5.o(.text+0x246): In >>function `error': >>/mad/minimad.c:177: undefined reference to `_mad_stream_errorstr' >>/cygdrive/c/DOCUME~1/USERNAME/LOCALS~1/Temp/ccyf7LL5.o(.text+0x2a7): In >>function `decode': >>/mad/minimad.c:209: undefined reference to `_mad_decoder_init' >>/cygdrive/c/DOCUME~1/USERNAME/LOCALS~1/Temp/ccyf7LL5.o(.text+0x2b2): >>/mad/minimad.c:215: undefined reference to `_mad_decoder_run' >>/cygdrive/c/DOCUME~1/USERNAME/LOCALS~1/Temp/ccyf7LL5.o(.text+0x2bc): >>/mad/minimad.c:219: undefined reference to `_mad_decoder_finish' >>collect2: ld returned 1 exit status >>make: *** [minimad] Error 1" >> >>I don't know what I'm doing wrong or even what it means by "undefined >>reference". > >Try instead: make minimad.exe > >-- >Rob Leslie >rob(a)mars.org > _________________________________________________________________ Download games, logos, wallpapers and lots more at MSN Mobile! http://www.msn.com.sg/mobile/

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Compiling minimad.c under Windows
by Some One 06 Jun '04

06 Jun '04

Hi, I'm trying to compile minimad.c using Cygwin and GCC. I've copied the "libmad-0.51.1b" folder into the Cygwin root as "mad" and followed all the steps in the "INSTALL" file (configure, make, make install). It does all that fine (makes the Makefile and config file, creates the objects in the "./deps" folder, installs the mad library in "usr/local/lib" and puts "mad.h" in "usr/local/include"). However, when a try "make minimad" in the "mad" folder, it returns: "gcc -Wall -march=i486 -g -O -fforce-mem -fforce-addr -fthread-jumps -fcse-follow -jumps -fcse-skip-blocks -fexpensive-optimizations -fregmove -fschedule-insns2 -fstrength-reduce minimad.c -o minimad /cygdrive/c/DOCUME~1/USERNAME/LOCALS~1/Temp/ccyf7LL5.o(.text+0xc3): In function `input': /mad/minimad.c:92: undefined reference to `_mad_stream_buffer' /cygdrive/c/DOCUME~1/USERNAME/LOCALS~1/Temp/ccyf7LL5.o(.text+0x246): In function `error': /mad/minimad.c:177: undefined reference to `_mad_stream_errorstr' /cygdrive/c/DOCUME~1/USERNAME/LOCALS~1/Temp/ccyf7LL5.o(.text+0x2a7): In function `decode': /mad/minimad.c:209: undefined reference to `_mad_decoder_init' /cygdrive/c/DOCUME~1/USERNAME/LOCALS~1/Temp/ccyf7LL5.o(.text+0x2b2): /mad/minimad.c:215: undefined reference to `_mad_decoder_run' /cygdrive/c/DOCUME~1/USERNAME/LOCALS~1/Temp/ccyf7LL5.o(.text+0x2bc): /mad/minimad.c:219: undefined reference to `_mad_decoder_finish' collect2: ld returned 1 exit status make: *** [minimad] Error 1" I don't know what I'm doing wrong or even what it means by "undefined reference". Thanks in advance for any help. _________________________________________________________________ Watch the online reality show Mixed Messages with a friend and enter to win a trip to NY http://www.msnmessenger-download.click-url.com/go/onm00200497ave/direct/01/

2 1

The madplay can't play mp3.
by kobi.yang＠billionton.com.tw 19 May '04

19 May '04

I have three mp3 songs and the madplay can't play it. Maybe, it is mp3's problem, but i can play those mp3 in the windows media player. Can any body help me to solve this problem. You can download mp3 songs below. Thanks for help. IP: 203.204.108.161 username: test password: 1234

2 1

lost synchronization and CRC check failed
by chikku belli 10 May '04

10 May '04

Hi all, I am new to audio and video decoder, hence would appreciate some help in the beginning. I have an mpg file which has both and audio and video stream encoded. I use libmpeg2 to decode the video and display it on the monitor. I also use libmad to play the audio. when i decode the mpg file i store audio part of the mpg file in a separate file (for ex: audio.ma2). After mpg file decoding is over, i run madplay with the following options. madplay -v /root/audio.ma2 I see the following error message on the screen. MPEG Audio Decoder 0.15.2 (beta) - Copyright Â© 2000-2004 Robert Leslie et al. error: frame 0: lost synchronization 00:00:00 Layer II, 224 kbps, 44100 Hz, stereo, CRC error: frame 3: lost synchronization 00:00:00 Layer II, 224 kbps, 44100 Hz, stereo, CRC error: frame 6: lost synchronization 00:00:00 Layer II, 224 kbps, 44100 Hz, stereo, CRC error: frame 9: lost synchronization 00:00:00 Layer II, 224 kbps, 44100 Hz, stereo, CRC error: frame 12: CRC check failed 00:00:00 Layer II, 224 kbps, 44100 Hz, stereo, CRC error: frame 14: lost synchronization 00:00:00 Layer II, 224 kbps, 44100 Hz, stereo, CRC error: frame 17: lost synchronization 00:00:00 Layer II, 224 kbps, 44100 Hz, stereo, CRC error: frame 20: lost synchronization 00:00:00 Layer II, 224 kbps, 44100 Hz, stereo, CRC error: frame 23: CRC check failed 00:00:00 Layer II, 224 kbps, 44100 Hz, stereo, CRC error: frame 25: lost synchronization 00:00:00 Layer II, 224 kbps, 44100 Hz, stereo, CRC error: frame 28: lost synchronization 00:00:00 Layer II, 224 kbps, 44100 Hz, stereo, CRC error: frame 31: lost synchronization 00:00:00 Layer II, 224 kbps, 44100 Hz, stereo, CRC error: frame 34: lost synchronization 00:00:00 Layer II, 224 kbps, 44100 Hz, stereo, CRC error: frame 36: lost synchronization 00:00:01 Layer II, 224 kbps, 44100 Hz, stereo, CRC error: frame 39: lost synchronization 00:00:01 Layer II, 224 kbps, 44100 Hz, stereo, CRC could anybody please tell me what could be the problem? regards, -onup Yahoo! India Matrimony: Find your partner online.

2 1

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