diff options
| -rw-r--r-- | README | 200 | ||||
| -rw-r--r-- | toolame.c | 6 |
2 files changed, 108 insertions, 98 deletions
@@ -5,7 +5,7 @@ ODR-DabMux compatible ZeroMQ output and JACK input Copyright (C) 2002, 2003 Michael Cheng [mikecheng at NOT planckenergy.com] remove the NOT http://www.planckenergy.com/ -Copyright (C) 2014 Matthias P. Braendli [matthias@mpb.li] +Copyright (C) 2014, 2015 Matthias P. Braendli [matthias@mpb.li] http://opendigitalradio.org/ All changes to the ISO source are licensed under the LGPL @@ -54,115 +54,123 @@ USAGE ./toolame [options] <input> <output> Input - tooLAME parses AIFF and WAV files for file info - raw PCM is assumed if no header is found - for stdin use a - - for JACK input, use -j option, and specify the name - of the JACK port with <input> + tooLAME parses AIFF and WAV files for file info + raw PCM is assumed if no header is found + for stdin use a - + for JACK input, use -j option, and specify the name + of the JACK port with <input> + for the libvlc based input, use -V, and specify the URL + with <input> Output - file is automatically renamed from *.* to *.mp2 - for stdout use a - + file is automatically renamed from *.* to *.mp2 + for stdout use a - for zeromq use tcp://<hostname>:<port> pointing to a ODR-DabMux Input Options - -s [int] - if inputting raw PCM sound, you must specify the sample rate - default sample rate is 44.1khz. + -s [int] + if inputting raw PCM sound, you must specify the sample rate + default sample rate is 44.1khz. - -a - downmix from stereo to mono - if the incoming file is stereo, combine the audio into - a single channel + -a + downmix from stereo to mono + if the incoming file is stereo, combine the audio into + a single channel - -x - force byte-swapping of the input. (current endian detection is dodgy, - so if toolame produces only noise, use -x ) + -x + force byte-swapping of the input. (current endian detection is dodgy, + so if toolame produces only noise, use -x ) - -g - swap the LR channels of a stereo file + -g + swap the LR channels of a stereo file Output Options - -m [char] - the encoding mode (default 'j') - 's' stereo - 'd' dual channel - 'j' joint stereo - 'm' mono - - -p [int] - which psy model to use (default '1') - Different models for the psychoacoustics - Models: -1 to 4 - - -b [int] - the total bitrate - For 48/44.1/32kHz default = 192 - For 24/22.05/16kHz default = 96 - - -v [int] - Switch on VBR mode. - The higher the number the better the quality. - Useful range -10 to 10. - See README.VBR for details. - Don't use that for DAB encoding. + -m [char] + the encoding mode (default 'j') + 's' stereo + 'd' dual channel + 'j' joint stereo + 'm' mono + + -p [int] + which psy model to use (default '1') + Different models for the psychoacoustics + Models: -1 to 4 + + -b [int] + the total bitrate + For 48/44.1/32kHz default = 192 + For 24/22.05/16kHz default = 96 + + -v [int] + Switch on VBR mode. + The higher the number the better the quality. + Useful range -10 to 10. + See README.VBR for details. + Don't use that for DAB encoding. Operation - -f - fast mode turns off calculation of the psychoacoustic model. - Instead a set of default values are assumed + -f + fast mode turns off calculation of the psychoacoustic model. + Instead a set of default values are assumed - -q [int] - quick mode calculates the psy model every 'num' frames. + -q [int] + quick mode calculates the psy model every 'num' frames. Misc - -d emp - de-emphasis (default 'n') - -c - mark as copyright - -o - mark as original - -e - add error protection - -r - force padding bits off - -D - add DAB extensions - -t [int] - 'talkativity' setting. 0 = no message. 3 = too much information + -d emp + de-emphasis (default 'n') + -c + mark as copyright + -o + mark as original + -e + add error protection + -r + force padding bits off + -D + add DAB extensions + -t [int] + 'talkativity' setting. 0 = no message. 3 = too much information ********************* EXAMPLES ********************* 1. - toolame sound.wav + toolame sound.wav - This will encode sound.wav to sound.mp2 using the default bitrate of 192 kbps - and using the default psychoacoustic model (model 1) + This will encode sound.wav to sound.mp2 using the default bitrate of 192 kbps + and using the default psychoacoustic model (model 1) 2. - toolame -p 2 -v 5 sound.wav newfile.mp2 + toolame -p 2 -v 5 sound.wav newfile.mp2 - Encode sound.wav to newfile.mp2 using psychoacoustic model 2 and encoding - with variable bitrate. The high value of the "-v" argument means that - the encoding will tend to favour higher bitrates. + Encode sound.wav to newfile.mp2 using psychoacoustic model 2 and encoding + with variable bitrate. The high value of the "-v" argument means that + the encoding will tend to favour higher bitrates. 3. - toolame -p 2 -v -5 sound.wav newfile.mp2 + toolame -p 2 -v -5 sound.wav newfile.mp2 - Same as example above, except that the negative value of the "-v" argument - means that the lower bitrates will be favoured over the higher ones. + Same as example above, except that the negative value of the "-v" argument + means that the lower bitrates will be favoured over the higher ones. 4. - cat sound.pcm | toolame -s 22050 -f -b 96 - newfile.mp2 + cat sound.pcm | toolame -s 22050 -f -b 96 - newfile.mp2 - Toolame is encoding from stdin at a bitrate of 96kbps and is using the - 'fast' mode which means that no psychoacoustic modelling is done.The + Toolame is encoding from stdin at a bitrate of 96kbps and is using the + 'fast' mode which means that no psychoacoustic modelling is done.The input file is raw pcm so the sample rate needs to be specified (22050Hz) +5. + toolame -V -s 48 -L -b 128 http://your_stream_url tcp://localhost:9002 + + Use libvlc to read and decode the stream at the given url, and encode it at + 48000Hz sample rate to the ZeroMQ input of ODR-DabMux running on localhost:9002 + ********************* CONTRIBUTORS @@ -170,48 +178,48 @@ CONTRIBUTORS Dist10 code writers LAME specific contributions - fht routines from Ron Mayer <mayer at acuson.com> - fht tweaking by Mathew Hendry <math at vissci.com> - window_subband & filter_subband from LAME circa v3.30 - (multiple LAME authors) - (before Takehiro's window/filter/mdct combination) + fht routines from Ron Mayer <mayer at acuson.com> + fht tweaking by Mathew Hendry <math at vissci.com> + window_subband & filter_subband from LAME circa v3.30 + (multiple LAME authors) + (before Takehiro's window/filter/mdct combination) Oliver Lietz <lietz at nanocosmos.de> - Tables now included in the exe! (yay! :) + Tables now included in the exe! (yay! :) Patrick de Smet <pds at telin.rug.ac.be> - scale_factor calc speedup. - subband_quantization speedup + scale_factor calc speedup. + subband_quantization speedup Federico Grau <grauf at rfa.org> Bill Eldridge <bill at hk.rfa.org> - option for "no padding" + option for "no padding" Nick Burch <gagravarr at SoftHome.net> - WAV file reading - os/2 Makefile mods. + WAV file reading + os/2 Makefile mods. Phillipe Jouguet <philippe.jouguet at vdldiffusion.com> - DAB extensions - spelling, LSF using psyII, WAVE reading + DAB extensions + spelling, LSF using psyII, WAVE reading Henrik Herranen - leopold at vlsi.fi - (WAVE reading) + (WAVE reading) Andreas Neukoetter - anti at webhome.de - (verbosity patch '-t' switch for transcode plugin) + (verbosity patch '-t' switch for transcode plugin) Sami Sallinen - sami.sallinen at g-cluster.com - (filter_subband loop unroll - psycho_i fix for "% 1408" calcs) + (filter_subband loop unroll + psycho_i fix for "% 1408" calcs) Mike Cheng <mikecheng at NOT planckenergy.com> (remove the NOT) - Most of the rest + Most of the rest Matthias P. Braendli <matthias@mpb.li> - ZeroMQ output for ODR-DabMux - PAD insertion for DLS and slides - JACK input + ZeroMQ output for ODR-DabMux + PAD insertion for DLS and slides + JACK and libVLC input ********************* REFERENCE PAPERS @@ -668,7 +668,8 @@ void usage (void) fprintf (stdout, "\t-a downmix from stereo to mono\n"); fprintf (stdout, "\t-x force byte-swapping of input\n"); fprintf (stdout, "\t-g swap channels of input file\n"); - fprintf (stdout, "\t-j enable jack input\n"); + fprintf (stdout, "\t-j use jack input\n"); + fprintf (stdout, "\t-V use libvlc input\n"); fprintf (stdout, "\t-L enable audio level display\n"); fprintf (stdout, "Output\n"); fprintf (stdout, "\t-m mode channel mode : s/d/j/m (dflt %4c)\n", @@ -720,7 +721,7 @@ void short_usage (void) fprintf (stderr, "tooLAME version %s\n (http://opendigitalradio.org)\n", toolameversion); fprintf (stderr, "MPEG Audio Layer II encoder for DAB\n\n"); - fprintf (stderr, "USAGE: %s [options] (<infile>|<jackname>) [outfile]\n\n", programName); + fprintf (stderr, "USAGE: %s [options] (<infile>|<jackname>|<libvlc url>) [outfile]\n\n", programName); fprintf (stderr, "Try \"%s -h\" for more information.\n", programName); exit (0); } @@ -737,6 +738,7 @@ void short_usage (void) * syntax: * * -j turns on JACK input + * -V turns on libVLC input * -L turns on audio level display * -m is followed by the mode * -y is followed by the psychoacoustic model number |