mad-dev July 2004

mad-dev@lists.mars.org

9 participants
15 discussions

mp3gain vs madplay
by Bruce Fitzsimons 15 Sep '04

15 Sep '04

Peoples, I'm about to launch into a debugging session to see why mp3gain (http://mp3gain.sourceforge.net/ ) and madplay 0.15.1b don't seem to be playing together (no pun intended). I thought I'd ask if anyone has used the two together yet, or if there is another linux-based replaygain processor (for adding replaygain to existing mp3s) that is more reliable. map3gain does appear to be adding a tag, as it finds it again if rerun over the same file. "# mp3gain /tmp/Abracadabra.mp3 /tmp/Abracadabra.mp3 Recommended "Track" dB change: -1.690000 Recommended "Track" mp3 gain change: -1 Max PCM sample at current gain: 28533.063680 Max mp3 global gain field: 180 Min mp3 global gain field: 116 Recommended "Album" dB change for all files: -7.490000 Recommended "Album" mp3 gain change for all files: -5" madplay doesn't appear to find it: "# madplay -G -Tv /tmp/Abracadabra.mp3 MPEG Audio Decoder 0.15.1 (beta) - Copyright (C) 2000-2004 Robert Leslie et al. Title: Abracadabra Artist: The Steve Miller Band Album: Golds Greatest Hits Track: 11 Year: 1982 Genre: Vocal Encoder Version: LAME 3.92 VBR Method: constant Bitrate: 255+ kbps Stereo Mode: normal Preset: none Unwise Settings: no Encoding Flags: Lowpass Filter: 19500 Hz ATH Type: 2 Noise Shaping: 1 Surround: none Start Delay: 576 samples End Padding: 1503 samples Source Rate: 44.1 kHz Music Length: 9948263 bytes Audio Frames: 11900" (note the missing "Replay Gain" printout I expect after reading the source) I'd hasten to add that I suspect mp3gain of doing something wrong, not madplay. I'm guessing the tag is wrong or in the wrong place. Any info would be appreciated, otherwise I'll report my findings to the list when I discover the root cause. /Bruce

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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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mad-dev July 2004