| Audio-FindChunks documentation | view source | Contained in the Audio-FindChunks distribution. |
Audio::FindChunks - breaks audio files into sound/silence parts.
use Audio::FindChunks;
# Duplicate input to output, caching RMS values to a file (as a side effect)
Audio::FindChunks->new(rms_filename => 'x.rms', filter => 1)->get('rms_data');
# Output human-readable info, using RMS cache file 'xxx.rms' if present:
Audio::FindChunks->new(cache_rms => 1, filename => 'xxx.mp3',
stem_strip_extension => 1)->output_blocks();
# Remove start/end silence (if longer than 0.2sec):
Audio::FindChunks->new(cache_rms => 1, filename => 'xxx.mp3',
min_actual_silence_sec => 1e100)->split_file();
# Split a multiple-sides tape recording
Audio::FindChunks->new(filename => 'xxx.mp3', min_actual_silence_sec => 11
)->split_file({verbose => 1});
# Output the RMS levels of small interval in human-readable form
Audio::FindChunks->new(filename => 'xxx.mp3')->output_levels();
Audio sequence is broken into parts which contain only noise ("gaps"), and parts with usable signal ("tracks").
The following configuration settings (and defaults) are supported:
# For getting PCM flow (and if averaging data is read from cache)
frequency => 44100, # If 'raw_pcm' or 'override_header_info' only
bytes_per_sample => 4, # likewise
channels => 2, # likewise
sizedata => MY_INF, # likewise (how many bytes of PCM to read)
out_fh => \*STDOUT, # mirror WAV/PCM to this FH if 'filter'
# Process non-WAV data:
preprocess => {mp3 => [[qw(lame --silent --decode)], [], ['-']]}, # Second contains extra args to read stdin
# RMS cache (used if 'valid_rms')
rms_extension => '.rms', # Appended to the 'filestem'
# Averaging to RMS info
sec_per_chunk => 0.1, # The window for taking mean square
# thresholds picking from the list of sorted 3-medians of RMS data
threshold_in_sorted_min_rel => 0, # relative position of 'threashold_min'
threshold_in_sorted_min_sec => 1, # shifted by this amount in the list
threshold_factor_min => 1, # the list elt is multiplied by this
threshold_in_sorted_max_rel => 0.5, # likewise
threshold_in_sorted_max_sec => 0, # likewise
threshold_factor_max => 1, # likewise
threshold_ratio => 0.15, # relative position between min/max
# Chunkification: smoothification
above_thres_window => 11, # in units of chunks
above_thres_window_rel => 0.25, # fractions of chunks above threshold
# in a window to make chunk signal
# Splitting into runs of signal/noise
max_tracks => 9999, # fail if more signal/noise runs
min_signal_sec => 5, # such runs of signal are forced
min_silence_sec => 2, # likewise
ignore_signal_sec => 1, # short runs of signal are ignored
min_silence_chunks_merge (see below) # and long resulting runs of silence
# are forced
# Calculate average signal in an interval "deeply inside" silence runs
local_level_ignore_pre_sec => 0.3, # offset the start of this interval
local_level_ignore_pre_rel => 0.02, # additional relative offset
local_level_ignore_post_sec => 0.3, # likewise for end of the interval
local_level_ignore_post_rel => 0.02, # likewise
# Enlargement of signal runs: attach consequent chunks with signal this much
# above this average over the neighbour silence run
local_threshold_factor => 1.05,
# Final enlargement of runs of signal
extend_track_end_sec => 0.5, # Unconditional enlargement
extend_track_begin_sec => 0.3, # likewise
min_boundary_silence_sec => 0.2, # Ignore short silence at start/end
Note that above_thres_window is the only value specified directly in
units of chunks; the other *_sec may be optionally specified in units
of chunks by setting the corresponding *_chunks value. Note also that
this window should better be decreased if minimal allowed silence length
parameters are decreased.
These values are mirrored from other values if not explicitly specified:
min_actual_silence_sec << min_silence_sec # Ignore short gaps min_start_silence_sec << min_boundary_silence_sec # Same at start min_end_silence_sec << min_boundary_silence_sec # Same at end min_silence_chunks_merge << min_silence_chunks # See above cache_rms_write <<< cache_rms # Boolean: write RMS cache cache_rms_read <<< cache_rms # Boolean: read RMS cache (unless 'filter')
The following values default to undef:
filename # if undef, read data from STDIN
stem_strip_extension # Boolean: 'filestem' has no extension
filter # If true, PCM data is mirrored to out_fh
rms_filename # Specify cache file explicitly
raw_pcm # The input has no WAV header
override_header_info # The user specified values override WAV header
cache_rms # Use cache file (see *_write, *_read above)
skip_medians # Boolean: do not calculate 3-medians
subchunk_size # Optimization of calculation of RMS; the
# best value depends on the processor cache
new(key1 => value1, key2 => value2, ....)The arguments form a hash of configuration parameters.
set(key => value)set a configuration parameter.
get(key)get a configuration parameter or a value which may be calculated basing on them.
output_levels([key])prints a human-readable display of RMS (or similar) values. Defaults to
rms_data; additional possible values are medians and sorted.
The format of the output data is similar to
Frequency: 44100. Stride: 4; 2 channels.
Chunk=0.1sec=17640bytes.
ch0: -9999.0 .. 9999.0 (-10dB;-10dB). ch1: -9999.0 .. 9999.0 (-10dB;-10dB).
0: 0.0: 20.7= -61dB: ###########>
1: 0.1: 20.7= -61dB: ###########>
2: 0.2: 20.7= -61dB: ###########>
...
(with the ch0 ETC line empty if data is read from an RMS file). Each
chunk gives a line with the chunk number, start (in sec), RMS intensity
(in linear scale and in decibel), and the graphical representation of the
decibel level (each # counts as 3dB, : adds 1dB, and >
adds 2dB).
output_blocks([option_hashref], [key])prints a human-readable display of obtained audio chunks. key defaults to
b; additional possible values are b0 to b4. Recognized options key
is format; defaults to long, which results in windy output; the value
short results in shorter output and no preamble. Preamble lines are all
#-commented; any output line is in the form
START_SEC =END_SEC # COMMENT
With short format there is no preamble, and (currently) COMMENT is of
the form PIECE_NUMBER len=PIECE_DURATION_SEC. These formats are
recognized, e.g., by MP3::Split::mp3split_read().
The default format is currently
# threshold: 1078.46653890971 (in 20.7214163971884 .. 7072.35556648067) 4.4 =25.8 # n=1 duration 21.4; gap 4.4 (4.4 .. 25.8; 21.4) 27.7 =67 # n=2 duration 39.3; gap 1.9 (27.7 .. 1m07.0; 39.3)
split_file([options], [key])Splits the file (only MP3 via MP3::Splitter is supported now). The
meaning of options is the same as for MP3::Splitter. Defaults to
blocks of type b; additional possible values are b0 to b4.
Duplicate RMS info between two different Audio::FindChunks objects.
The exchanged info is the following:
chunks rms_data medians sorted channels min max
frequency bytes_per_sample sec_per_chunk bytes_per_chunk
set_rmsinfo() returns the object itself.
The functionality of the module is modelled on the architecture of
Data::Flow: the two principal methods are set(key => value)
and get(key); the module knows how to calculate keys basing on values of
other keys.
The results of calculation are cached; in particular, if one needs to calculate
some value for different values of a configuration parameter, one should
create many copies of Audio::FindChunks object, as in
my @info = Audio::FindChunks->new(filename => $f)->get_rmsinfo;
for my $ratio (0..100) {
Audio::FindChunks->new(threshold_ratio => $r/100)
->set_rmsinfo(@info)->print_blocks();
}
The internally used format of intermediate data is designed for quick shallow copying even for enourmous audio files.
The current dependecies for values which are not explicitly set():
filestem <<< filename stem_strip_extension input_type <<< filename preprocess_a <<< input_type preprocess preprocess_input <<< preprocess_a filename fh AND close_fh <<< preprocess_input filename fh_bin <<< fh out_fh_bin <<< filter out_fh rms_filename_default <<< filestem rms_extension read_from_rms_file <<< filter cache_rms_read rms_filename write_to_rms_file <<< cache_rms_write rms_filename rms_filename_actual <<< rms_filename rms_filename_default samples_per_chunk <<< sec_per_chunk frequency bytes_per_chunk <<< samples_per_chunk bytes_per_sample rms_data_arr_f <<< read_from_rms_file rms_filename_actual samples_per_chunk rms_data AND chunks <<< rms_data_arr_f OR A LOT OF OTHER PARAMETERS medians <<< rms_data skip_medians chunks sorted <<< medians chunks, threshold_in_sorted_* <<< chunks threshold_in_sorted_*_* threshold_min/max <<< threshold_factor_* sorted threshold_in_sorted_min/max threshold <<< threshold_min threshold_ratio threshold_max above_thres <<< chunks rms_data threshold above_thres_in_window <<< above_thres chunks above_thres_window above_thres_window_abs<<< above_thres_window_rel above_thres_window maybe_signal <<< above_thres_in_window chunks above_thres_window_abs maybe_trk_pk <<< max_tracks maybe_signal chunks b0 <<< maybe_trk_pk b1 <<< b0 min_signal_chunks min_silence_chunks b2 <<< b1 ignore_signal_chunks b3 <<< b2 min_silence_chunks_merge b4 <<< b3 b <<< b4 local_level_ignore_* medians local_threshold_factor extend_track_begin_chunks extend_track_end_chunks min_actual_silence_chunks min_start_silence_chunks min_end_silence_chunks
If rms_data is not read from cached source, a lot of other fields may
be also set from the WAV header (unless raw_pcm).
Potentially large internally-cached values are stored as array references to decrease the overhead of shallow copying.
The data which relates to
the initial chunks (of size sec_per_chunk) is stored as length 1 arrays
with packed (either by l* or d*, depending on the semantic) data; this
allows small memory footprint work with huge audio files, and allows
an easy implemenation of most computationally intensive work in C.
The blocks of audio/signal/noise/silence are stored as Perl arrays; each element is a reference to an array of length 3: type (-1 for silence, 0 for noise, 1 for signal, and 2 for audio), start chunks, duration in chunks.
The algorithm for finding boundaries of parts follows closely the algorithm
used by GramoFile v1.7 (however, this version is fully customizable,
fully documented, and has some significant bugs fixed). The keywords in the
discussion below refer to customization parameters; keywords of the form
>>>key refer to get()able values set on the step in
question.
This is done in 2 distinct steps:
Break the input into chunks of equal duration (governed by sec_per_chunk);
find the acoustic energy of each channel per chunk (no customization);
energy is the quadratic average of signal level; calculate maximal
energy among channels per chunk (no customization; >>>rms_data).
Trim "extremal" chunks by replacing the energy level of each chunk by
the median of it and its two neighbors (switched off if skip_medians;
>>>medians).
basing on the distribution (>>>sorted) of smoothed values.
Governed by threshold_* parameters. >>>threshold_min,
>>>threshold_max, >>>threshold.
Separate into signal and noise chunks basing on the number of
above-threshold chunks in a small window about the given chunk. Governed by
above_thres_window, above_thres_window_rel. >>>maybe_signal,
>>>b0.
Long enough runs of signal chunks are proclaimed carrying sound; likewise
for noise chunks and silence. Governed by max_tracks, min_signal_chunks,
min_silence_chunks. >>>b1.
Long enough "unproclaimed" runs of chunks with only short bursts of
signal are proclaimed silence. Governed by ignore_signal_chunks,
>>>b2; and min_silence_chunks_merge, >>>b3.
A run of chunks (signal or noise) "yet unproclaimed" to be sound or silence is proclaimed sound if it is adjacent to a run of sound on at least one side. The rest of unproclaimed runs are proclaimed silence. No customization.
Runs of sound/silence are audio/gap candidates (no customization;
>>>b4).
ignoring short intervals near ends of gaps. Governed by local_level_*.
Extend runs of audio: join the consequent runs of chunks of adjacent gaps
where the energy level
remains significantly larger than the average level in this gap.
Additionally, unconditionally extend the tracks by a small amount.
Governed by local_threshold_factor, extend_track_end_chunks,
extend_track_begin_chunks.
Gaps which became too short are considered audio and are merged into
neighbors. Governed by min_actual_silence_chunks, min_start_silence_chunks,
min_end_silence_chunks; >>>b.
long bool_find_runs(int *input, array_run_t *output, long cnt, long out_cnt) void double_find_above(double *input, int *output, long cnt, double threshold) void double_median3(double *rmsarray, double *medarray, long total_blocks) void double_sort(double *input, double *output, long cnt) void int_find_above(int *input, int *output, long cnt, int threshold) void int_sum_window(int *input, int *output, long cnt, int window_size) void le_short_sample_stats(char *buf, int stride, long samples, array_stats_t *stat)
Data::Flow, MP3::Split
Ilya Zakharevich, <cpan@ilyaz.org<gt>
Copyright (C) 2004 by Ilya Zakharevich
This library is free software; you can redistribute it and/or modify it under the same terms as Perl itself, either Perl version 5.8.2 or, at your option, any later version of Perl 5 you may have available.
| Audio-FindChunks documentation | view source | Contained in the Audio-FindChunks distribution. |