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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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