mad-dev

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682 discussions

ReplayGain in RVA2 frames
by Joris van Rantwijk 24 May '09

24 May '09

Hello, I noticed that Madplay supports ReplayGain, which is nice because ReplayGain is a very useful idea. However, Madplay currently does not read ReplayGain information from ID3v2 RVA2 frames. (Madplay does read RVA2 frames, but doesn't look at the identification string which distinguishes between track-gain and album-gain.) This is unfortunate because it seems that, out of at least five distinct methods for storing ReplayGain information, RVA2 is now best supported among audio players. I would like to make the following changes to the ReplayGain code in madplay: * Support ReplayGain from RVA2 frames in addition to RGAD frames. This will improve compatibility with other audio players, making ReplayGain much more usable. * Remove the +6 dB pre-amplification thing. The original ReplayGain standard specified normalize to 83 dB and amplify +6 dB. But this was later revised into simply normalize to 89 dB, which is how all current implementations are doing it. The +6 dB could be left in place for LAME tags older than 3.95.1. I am willing to write and test a patch that implements these changes. Would you accept that patch? Greetings, Joris van Rantwijk.

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How to get the accurate song time?
by Carlos Henrique 06 May '09

06 May '09

Hi to all, I neet to get the time of the song when it is playing. I am using the code below: static enum mad_flow header(void *data, struct mad_header const *header) { mad_timer_add(&musicTimer, header->duration); return MAD_FLOW_CONTINUE; } long mp3_song_time(void) { return mad_timer_count(musicTimer, MAD_UNITS_MILLISECONDS); } The code works, but it is not fast enough. I am getting the time with " mp3_song_time" faster than the header is called. (obs: I am using two processors, one to decode the song and the other to run the code that uses the song time). Is there another way to get the right (more accurate) time? Thanks in advance, -- Carlos Henrique Gomes Correia Game Programmer Interama Games Skype: carloshgcorreia Tel: +55 27 3345 8525 Web: www.interama.net

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Maintaining libid3tag?
by Andy Grundman 02 Apr '09

02 Apr '09

Is there any interest in having someone take over maintenance of libid3tag? There are several patches floating around in various places [1] that should be rolled into an official release. There are also some issues related to handling of broken iTunes tags that I'd like to fix and don't see any patches for. I'm happy to volunteer. [1] http://patch-tracking.debian.net/package/libid3tag http://bugs.gentoo.org/show_bug.cgi?id=210564 -- Andy Grundman Logitech Streaming Media Systems (Slim Devices) andy(a)slimdevices.com / www.slimdevices.com

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About one application for decoder design
by leon deng 22 Jan '09

22 Jan '09

Hi,all How are you? I found a source code for mp3 decorder using libmad library. It is called MADAudioFormatReader. if you do not know, it is not a big deal. In this code, It can handle the stereo (2 channles )mp3 file, However, it can not handle single channel mp3 file. What's the decorder difference between one channel and two channels mp3 file? I do not think there is a big difference. I think the key is to know the structure in data chunks for both case and use the right function in MAD. Am I right? Could you give me some clue? I attach the file to this email for reference. Best Leon

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About Xing VBR tag data
by leon deng 14 Jan '09

14 Jan '09

Hi ,all I am new to MAD. I got confused about XING tag. Is XING also a mp3 encoder? or XING is Tag in the mp3 format. If it is the later case, in the encoder side, we can put the size of the data in XING tag, and we can get the size of the file in XING tag in the decoder side. Because, we do not want to calculate the data size,and want to get the data size in XING tag. I also found XING is also a mp3 encoder on website. So could you tell me what the XING is ?

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How to achieve mp3 streaming using MAD?
by leon deng 13 Jan '09

13 Jan '09

Hi,all I want to find some mp3 headers, such as XING header,which can provide the duration information(total play time). I want to make use of it to achieve mp3 streaming support in libmad. So there are two questions. 1.I wrote a mp3 header reader, and it reads headers in the first 10 frames. I only read first 4 bytes for each frame. where is XING duration field located, or could you provide some header documentation for XING header to me? 2. Could you please tell me some other headers which can provide duration information in it? Best, Leon

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playback/decoding problem
by Mathias Kuester 02 Dec '08

02 Dec '08

Hello, i have a problem with one stream that is not correct decoded by libmad. I use libmad-0.15.1b-r5 and mplayer with the following command line: mplayer -ac mad http://gffstream.ic.llnwd.net/stream/gffstream_w19a This is a mp3 radio stream and i get ugly noise and spikes. I can try another stream and it works very well. Can anyone tell me why this stream is not correct decoded. I have tried some other decoders but its always the same problem. Thanks, Mathias --- OUTPUT: mplayer -ac mad http://gffstream.ic.llnwd.net/stream/gffstream_w19a Connecting to server gffstream.ic.llnwd.net[87.248.219.36]: 80... Cache size set to 320 KBytes Cache fill: 15.00% (49152 bytes) Audio only file format detected. ========================================================================== Forced audio codec: mad Opening audio decoder: [libmad] libmad mpeg audio decoder AUDIO: 44100 Hz, 2 ch, s16le, 128.0 kbit/9.07% (ratio: 16000->176400) Selected audio codec: [mad] afm: libmad (libMAD MPEG layer 1-2-3) ========================================================================== AO: [alsa] 48000Hz 2ch s16le (2 bytes per sample) Video: no video Starting playback... ---

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Playback issues
by Henrik Andersson 26 Oct '08

26 Oct '08

Hi, I am trying to make a sound library for the Wii that uses libmad for the mp3 decoding. However, I seem to have a problem, my code reports a lot of lost sync errors when decoding the frames. Could someone please take a look at my code and see if I am doing something obviously wrong? You can find my code at svn://henke37.cjb.net/sound/ , the main file is newMP3Player.cpp .

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Madplay improved patch for alsa audio
by Micha Nelissen 19 Oct '08

19 Oct '08

Hi, Attached is an improved patch for alsa backend of madplay. * fix allocation of buffer, buffer_size is number of frames * use default 16 bit depth (some hardware has small buffers, 16/24 bit difference is very small) Regards, Micha --- madplay-0.15.2b/audio_alsa.c 2008-10-18 15:10:16.000000000 +0200 +++ madplay-0.15.2b/audio_alsa.c.new 2008-10-18 15:03:27.000000000 +0200 @@ -28,31 +28,30 @@ #include <errno.h> -#define ALSA_PCM_OLD_HW_PARAMS_API -#define ALSA_PCM_OLD_SW_PARAMS_API #include <alsa/asoundlib.h> #include <mad.h> #include "audio.h" -char *buf = NULL; -int paused = 0; +#define BUFFER_TIME_MAX 500000 -int rate = -1; -int channels = -1; -int bitdepth = -1; -int sample_size = -1; - -int buffer_time = 500000; -int period_time = 100000; -char *defaultdev = "plughw:0,0"; +unsigned char *buf = NULL; +int paused = 0; + +unsigned int rate = 0; +unsigned int channels = -1; +unsigned int bitdepth = -1; +unsigned int sample_size = -1; + +unsigned int buffer_time; +unsigned int period_time; +char *defaultdev = "plughw:0,0"; snd_pcm_hw_params_t *alsa_hwparams; snd_pcm_sw_params_t *alsa_swparams; -snd_pcm_sframes_t buffer_size; -snd_pcm_sframes_t period_size; +snd_pcm_uframes_t buffer_size; snd_pcm_format_t alsa_format = -1; snd_pcm_access_t alsa_access = SND_PCM_ACCESS_MMAP_INTERLEAVED; @@ -66,14 +65,20 @@ snd_pcm_hw_params_t *params, snd_pcm_access_t access) { - int err, dir; - + int err; + /* choose all parameters */ err = snd_pcm_hw_params_any(handle,params); if (err < 0) { printf("Access type not available for playback: %s\n", snd_strerror(err)); return err; } + /* set the access type */ + err = snd_pcm_hw_params_set_access(handle, params, alsa_access); + if (err < 0) { + printf("Sample format not available for playback: %s\n", snd_strerror(err)); + return err; + } /* set the sample format */ err = snd_pcm_hw_params_set_format(handle, params, alsa_format); if (err < 0) { @@ -87,29 +92,38 @@ return err; } /* set the stream rate */ - err = snd_pcm_hw_params_set_rate_near(handle, params, rate, 0); + err = snd_pcm_hw_params_set_rate(handle, params, rate, 0); if (err < 0) { printf("Rate %iHz not available for playback: %s\n", rate, snd_strerror(err)); return err; } - if (err != rate) { - printf("Rate doesn't match (requested %iHz, get %iHz)\n", rate, err); - return -EINVAL; - } + err = snd_pcm_hw_params_get_buffer_time_max(params, &buffer_time, NULL); + if (err < 0) { + printf("Unable to retrieve buffer time: %s\n", snd_strerror(err)); + return err; + } + if (buffer_time > BUFFER_TIME_MAX) + buffer_time = BUFFER_TIME_MAX; /* set buffer time */ - err = snd_pcm_hw_params_set_buffer_time_near(handle, params, buffer_time, &dir); + err = snd_pcm_hw_params_set_buffer_time_near(handle, params, &buffer_time, 0); if (err < 0) { printf("Unable to set buffer time %i for playback: %s\n", buffer_time, snd_strerror(err)); return err; } - buffer_size = snd_pcm_hw_params_get_buffer_size(params); + if (period_time * 4 > buffer_time) + period_time = buffer_time / 4; /* set period time */ - err = snd_pcm_hw_params_set_period_time_near(handle, params, period_time, &dir); + err = snd_pcm_hw_params_set_period_time_near(handle, params, &period_time, NULL); if (err < 0) { printf("Unable to set period time %i for playback: %s\n", period_time, snd_strerror(err)); return err; } - period_size = snd_pcm_hw_params_get_period_size(params, &dir); + /* retrieve buffer size */ + err = snd_pcm_hw_params_get_buffer_size(params, &buffer_size); + if (err < 0) { + printf("Unable to retrieve buffer size: %s\n", snd_strerror(err)); + return err; + } /* write the parameters to device */ err = snd_pcm_hw_params(handle, params); if (err < 0) { @@ -123,6 +137,7 @@ int set_swparams(snd_pcm_t *handle, snd_pcm_sw_params_t *params) { + unsigned int start_threshold; int err; /* get current swparams */ @@ -136,13 +151,7 @@ if (err < 0) { printf("Unable to set start threshold mode for playback: %s\n", snd_strerror(err)); return err; - } - /* allow transfer when at least period_size samples can be processed */ - err = snd_pcm_sw_params_set_avail_min(handle, params, period_size); - if (err < 0) { - printf("Unable to set avail min for playback: %s\n", snd_strerror(err)); - return err; - } + } /* align all transfers to 1 samples */ err = snd_pcm_sw_params_set_xfer_align(handle, params, 1); if (err < 0) { @@ -190,7 +199,7 @@ rate = config->speed; if ( bitdepth == 0 ) - config->precision = bitdepth = 32; + config->precision = bitdepth = 16; switch (bitdepth) { @@ -241,7 +250,7 @@ return -1; } - buf = malloc(buffer_size); + buf = malloc(buffer_size * sample_size); if (buf == NULL) { audio_error="unable to allocate output buffer table"; return -1; @@ -279,7 +288,7 @@ int play(struct audio_play *play) { int err, len; - char *ptr; + unsigned char *ptr; ptr = buf; len = play->nsamples;

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libmad-0.15.1b does not compile on Fedora 9
by Joao Moreira 11 Oct '08

11 Oct '08

Hi, I've just installed a clean Fedora 9 distro, and I tried to compile libmad-0.15.1b (which I had done fine previously, under Fedora 5), I got the following error : gcc -DHAVE_CONFIG_H -I. -I. -I. -DFPM_INTEL -DASO_ZEROCHECK -Wall -march=i486 -g -O -fforce-mem -fforce-addr -fthread-jumps -fcse-follow-jumps -fcse-skip-blocks -fexpensive-optimizations -fregmove -fschedule-insns2 -fstrength-reduce -MT version.lo -MD -MP -MF .deps/version.Tpo -c version.c -fPIC -DPIC -o .libs/version.o cc1: error: unrecognized command line option "-fforce-mem" make[2]: *** [version.lo] Error 1 gcc version 4.3.0 does not have the "-fforce-mem" option... what can I do ? is there a way (a configure option) to disable this ? Thx, Joao -- Joao Moreira de Sa Coutinho joao.moreira(a)free.fr Tel.: 08.70.23.48.29 Mob.: 06.28.90.83.67

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