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292 discussions

mad decoder sample rates
by Steve Spano 07 Sep '07

07 Sep '07

Hi Folks, We have the mad library running on the new STR912 processor; 96MHz ARM9 core. We are using a small CPLD to bridge between the SSP serial port and a CS4431 stereo dac. We have an I2C silicon oscillator being used so we can set any sample rate we want. Currently, its working at 22.05KHz. We haven't change the sample rate yet. I want to be able to run 8Khz to 48KHz MP3s. Is there anything in the mad decoder that would prevent decoding MP3s within this range? We don't have a sample issue with our hardware since we can set any sample rate we want. Thanks Steve Spano, President steve(a)fl-eng.com Finger Lakes Engineering 607-277-1614 x223 www.fl-eng.com <http://www.fl-eng.com/>

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Compiling LIBMAD for ARM
by Karthika 02 Apr '07

02 Apr '07

Helloo, I am trying to compile LIBMAD for ARM7TDMI on Metrowerks Codewarrior tool. The processor specific options for ARM are present in the file fixed.h. I have taken those options which are specific for ARM and included them in a new header file called "asm.h" ( given below).I have included this "asm.h" file in my project. Then, I have removed fixed.h from my project. Now, when I compile my project I get the errors for mad_f_mul and MAD_F_MLA. MAD_F_MLA error is only for III_alias reduce( ) function. A few errors are given below. Error : #29: expected an expression layer3.c line 934 return frac ? mad_f_mul(requantized, root_table[3 + frac]) : requantized; Error : #18: expected a ")" layer3.c line 1591 MAD_F_MLA(hi, lo, -b, ca[i]); Error : #18: expected a ")" layer3.c line 1596 MAD_F_MLA(hi, lo, a, ca[i]); Error : #29: expected an expression layer3.c line 1647 m0 = mad_f_mul(a17, -c3); The "asm.h" file I have included is as follows. # define MAD_F_SCALEBITS 28 # define MAD_F_FRACBITS 28 # define MAD_F(x) ((mad_fixed_t) (x##L)) # define MAD_F_MIN ((mad_fixed_t) -0x80000000L) # define MAD_F_MAX ((mad_fixed_t) +0x7fffffffL) # define MAD_F_ONE MAD_F(0x10000000) # define mad_f_tofixed(x) ((mad_fixed_t) \ ((x) * (double) (1L << MAD_F_FRACBITS) + 0.5)) # define mad_f_todouble(x) ((double) \ ((x) / (double) (1L << MAD_F_FRACBITS))) # define mad_f_intpart(x) ((x) >> MAD_F_FRACBITS) # define mad_f_fracpart(x) ((x) & ((1L << MAD_F_FRACBITS) - 1)) /* (x should be positive) */ # define mad_f_fromint(x) ((x) << MAD_F_FRACBITS) # define mad_f_add(x, y) ((x) + (y)) # define mad_f_sub(x, y) ((x) - (y)) # define mad_f_mul(x, y) \ ({ mad_fixed64hi_t __hi; \ mad_fixed64lo_t __lo; \ mad_fixed_t __result; \ asm ("smull %0, %1, %3, %4\n\t" \ "movs %0, %0, lsr %5\n\t" \ "adc %2, %0, %1, lsl %6" \ : "=&r" (__lo), "=&r" (__hi), "=r" (__result) \ : "%r" (x), "r" (y), \ "M" (MAD_F_SCALEBITS), "M" (32 - MAD_F_SCALEBITS) \ : "cc"); \ __result; \ }) # define MAD_F_MLX(hi, lo, x, y) \ asm ("smull %0, %1, %2, %3" \ : "=&r" (lo), "=&r" (hi) \ : "%r" (x), "r" (y)) # define MAD_F_MLA(hi, lo, x, y) \ asm ("smlal %0, %1, %2, %3" \ : "+r" (lo), "+r" (hi) \ : "%r" (x), "r" (y)) # define MAD_F_MLN(hi, lo) \ asm ("rsbs %0, %2, #0\n\t" \ "rsc %1, %3, #0" \ : "=r" (lo), "=r" (hi) \ : "0" (lo), "1" (hi) \ : "cc") # define mad_f_scale64(hi, lo) \ ({ mad_fixed_t __result; \ asm ("movs %0, %1, lsr %3\n\t" \ "adc %0, %0, %2, lsl %4" \ : "=&r" (__result) \ : "r" (lo), "r" (hi), \ "M" (MAD_F_SCALEBITS), "M" (32 - MAD_F_SCALEBITS) \ : "cc"); \ __result; \ }) mad_fixed_t mad_f_abs(mad_fixed_t); mad_fixed_t mad_f_div(mad_fixed_t, mad_fixed_t); Can anybodu help me? Thanks, Karthika Lakshmi. __________________________________________________________ Yahoo! India Answers: Share what you know. Learn something new http://in.answers.yahoo.com/

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Synthesis filter bank in LIBMAD
by Karthika 28 Mar '07

28 Mar '07

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 coefficeints 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/

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Help :)
by Khalid Aallouche 09 Dec '06

09 Dec '06

Hi all, I wanted to decode an MP3 file in order to read it. But I'm novice and there is so much file...Is someone can help to create a basic MP3 player (console application:user enters where the MP3 is, and after read the valid MP3 file) Thank you very much Khalid

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madplay vs. mpg123
by Aaron 05 Dec '06

05 Dec '06

I ran more tests. madplay is fine on Mac. I used both madplay and mpg123 to decompress a file to wav. The wav sounds the same on both outputs. It sounds fine from aplay. It sounds bad (again) from audacity. It seems that on this machine, different programs/device drivers yield very different results. Any ideas what is different between these programs? (which drivers to they use... /dev/dsp? some alsa driver?) Perhaps there is some ioctl or something that isn't being done properly for my card on some programs... aaron

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Playing Internet Streams with Madplay
by Mark Kane 12 Oct '06

12 Oct '06

Hi everyone. This may be a stupid question, but I've read through the madplay man page and online documentation several times and I cannot get an MP3 Internet radio station (SHOUTcast) to play with madplay. Every time I try on the command line, I get: ------------------------------------------------------ [mixx941@amd64:~]% madplay http://stream.com:8000 MPEG Audio Decoder 0.15.2 (beta) - Copyright (C) 2000-2004 Robert Leslie et al. http://stream.com:8000: No such file or directory ------------------------------------------------------ I've tried with and without the trailing slash on the URL, with/without quotes on the URL, and adding the "listen.pls" at the end. Still the same. Tried downloading the .pls and playing it from the file, although that gives this: ------------------------------------------------------ [mixx941@amd64:~]% madplay listen.pls MPEG Audio Decoder 0.15.2 (beta) - Copyright (C) 2000-2004 Robert Leslie et al. error: frame 0: lost synchronization 0 frames decoded (0:00:00.0), -inf dB peak amplitude, 0 clipped samples [mixx941@amd64:~]% ------------------------------------------------------ What am I doing wrong? Thanks in advance. -Mark madplay-0.15.2b -- Internet Radio: Party107 (Trance/Electronic) - http://www.party107.com Rock 101.9 The Edge (Rock) - http://www.rock1019.net IRC: MIXXnet IRC Network - irc.mixxnet.net (Nick: MIXX941)

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Decoding time of MAD
by benix samuel 12 Oct '06

12 Oct '06

Hi everyone , I am using MAD decoder. It takes different time to decode two different frames of same header information and frame length. Is that MAD decoding time even depends on the data in the frame or only on the frame size. Benix

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Where the hell is definition of struct id3_file????
by tj 29 Sep '06

29 Sep '06

In trying to learn how to use libid3, since there sems to be no documentatio. I am looking at soure trying to figure out how to use it. The first problem is the practically every call uses a pointer to a "struct id3_file" . Now, the problem is that there is no definition for id3_file structure in any .h file. I finally found it in file.c, a source file. How the hell are you supposed to use it in your app when the prototype is in a package source file? Seems that something is really broken here. tj

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Madplay & Psion 5 MX: works 95% :-)
by Michael Fritscher 11 Sep '06

11 Sep '06

Hi, first of all: Thanks for the great program :-) madplay works ca. 95% on the psion 5 mx: it would need ca. 5-10% more power... It's a 36,8 Mhz ARM, the sound chip can do 8kHz/16 bit. I used the parameter --downscale, which scales 44.1 kHz mp3s down to 22.1 kHz, which helps much. But ideal would be a --downscale4 or so, which would scale it down to 11.05 kHz... How can I add this? I don't find the location in the code :-( Thanks, Michael Fritscher

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Gapless decoding with LAME tag reading
by Tobias Rundström 11 Aug '06

11 Aug '06

Hello List, I am trying to get our mad decoder in XMMS2 to playback LAME encoded files gapless by reading the decoder delay in the header, but I have some problems to get it working 100%. The test cases I have is confirmed playing gapless in foobar2000 so I guess I just doing something wrong :-) I parse the LAME tag with more or less the same code as in madplay.c and then I do: data->samples_to_skip = lame->start_delay; data->samples_to_skip_end = lame->end_padding; data->samples_to_play = (guint64)(((guint64)xmms_xing_get_frames (data->xing) * (guint64)1152) - (guint64)lame->start_delay); XMMS_DBG ("Samples to skip in the beginning: %d, total: %lld", data- >samples_to_skip, data->samples_to_play); These flags are then used in my decoding function which is attached at the bottom. Still I get around 1000 samples of silence in the beginning when I watch the output wave in a wave editor. My theories are that: 1) maybe MAD is adding silence to the output, I read something like hinted towards this on the interwebb somewhere. 2) that my find first sync code is not working properly. I feed the first frame (that I find the Xing/LAME tag in) back to the decoding routine found below, can that produce silence also? 3) My math is off somewhere :-) Any help on getting this play "perfect" gapless would be very appreciated! The full code is available here: http://git.xmms.se/? p=xmms2-devel.git;a=tree;;f=src/plugins/mad -- Tobias static gint xmms_mad_read (xmms_xform_t *xform, gpointer buf, gint len, xmms_error_t *err) { xmms_mad_data_t *data; xmms_samples16_t *out = (xmms_samples16_t *)buf; gint ret; gint j; gint read = 0; data = xmms_xform_private_data_get (xform); j = 0; while (read < len) { /* use already synthetized frame first */ if (data->synthpos < data->synth.pcm.length) { if (data->samples_to_skip <= 0 && data->samples_to_play >= 0) { out[j++] = scale_linear (data->synth.pcm.samples[0] [data->synthpos]); if (data->channels == 2) { out[j++] = scale_linear (data->synth.pcm.samples [1][data->synthpos]); read += 2 * xmms_sample_size_get (XMMS_SAMPLE_FORMAT_S16); data->samples_to_play -= 2; } else { read += xmms_sample_size_get (XMMS_SAMPLE_FORMAT_S16); data->samples_to_play --; } } else { if (data->channels == 2) { data->samples_to_skip -= 2; } else { data->samples_to_skip --; } } data->synthpos++; continue; } /* then try to decode another frame */ if (mad_frame_decode (&data->frame, &data->stream) != -1) { mad_synth_frame (&data->synth, &data->frame); data->synthpos = 0; continue; } /* if there is no frame to decode stream more data */ if (data->stream.next_frame) { guchar *buffer = data->buffer; const guchar *nf = data->stream.next_frame; memmove (data->buffer, data->stream.next_frame, data->buffer_length = (&buffer[data- >buffer_length] - nf)); } ret = xmms_xform_read (xform, (gchar *)data->buffer + data- >buffer_length, 4096 - data->buffer_length, err); if (ret <= 0) { return ret; } data->buffer_length += ret; mad_stream_buffer (&data->stream, data->buffer, data- >buffer_length); } return read; }

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