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author Guido van Rossum <guido@python.org>
date Thu, 17 Jul 1997 16:29:42 +0000
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1 \section{Built-in Module \sectcode{audioop}}
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2 \bimodindex{audioop}
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3
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4 The \code{audioop} module contains some useful operations on sound fragments.
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5 It operates on sound fragments consisting of signed integer samples
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6 8, 16 or 32 bits wide, stored in Python strings. This is the same
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7 format as used by the \code{al} and \code{sunaudiodev} modules. All
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8 scalar items are integers, unless specified otherwise.
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9
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10 A few of the more complicated operations only take 16-bit samples,
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11 otherwise the sample size (in bytes) is always a parameter of the operation.
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12
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13 The module defines the following variables and functions:
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14
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15 \renewcommand{\indexsubitem}{(in module audioop)}
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16 \begin{excdesc}{error}
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17 This exception is raised on all errors, such as unknown number of bytes
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18 per sample, etc.
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19 \end{excdesc}
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20
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21 \begin{funcdesc}{add}{fragment1\, fragment2\, width}
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22 Return a fragment which is the addition of the two samples passed as
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23 parameters. \var{width} is the sample width in bytes, either
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24 \code{1}, \code{2} or \code{4}. Both fragments should have the same
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25 length.
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26 \end{funcdesc}
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27
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28 \begin{funcdesc}{adpcm2lin}{adpcmfragment\, width\, state}
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29 Decode an Intel/DVI ADPCM coded fragment to a linear fragment. See
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30 the description of \code{lin2adpcm} for details on ADPCM coding.
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31 Return a tuple \code{(\var{sample}, \var{newstate})} where the sample
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32 has the width specified in \var{width}.
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33 \end{funcdesc}
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34
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35 \begin{funcdesc}{adpcm32lin}{adpcmfragment\, width\, state}
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36 Decode an alternative 3-bit ADPCM code. See \code{lin2adpcm3} for
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37 details.
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38 \end{funcdesc}
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39
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40 \begin{funcdesc}{avg}{fragment\, width}
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41 Return the average over all samples in the fragment.
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42 \end{funcdesc}
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43
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44 \begin{funcdesc}{avgpp}{fragment\, width}
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45 Return the average peak-peak value over all samples in the fragment.
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46 No filtering is done, so the usefulness of this routine is
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47 questionable.
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48 \end{funcdesc}
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49
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50 \begin{funcdesc}{bias}{fragment\, width\, bias}
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51 Return a fragment that is the original fragment with a bias added to
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52 each sample.
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53 \end{funcdesc}
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54
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55 \begin{funcdesc}{cross}{fragment\, width}
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56 Return the number of zero crossings in the fragment passed as an
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57 argument.
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58 \end{funcdesc}
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59
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60 \begin{funcdesc}{findfactor}{fragment\, reference}
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61 Return a factor \var{F} such that
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62 \code{rms(add(fragment, mul(reference, -F)))} is minimal, i.e.,
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63 return the factor with which you should multiply \var{reference} to
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64 make it match as well as possible to \var{fragment}. The fragments
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65 should both contain 2-byte samples.
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66
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67 The time taken by this routine is proportional to \code{len(fragment)}.
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68 \end{funcdesc}
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69
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70 \begin{funcdesc}{findfit}{fragment\, reference}
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71 This routine (which only accepts 2-byte sample fragments)
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72
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73 Try to match \var{reference} as well as possible to a portion of
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74 \var{fragment} (which should be the longer fragment). This is
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75 (conceptually) done by taking slices out of \var{fragment}, using
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76 \code{findfactor} to compute the best match, and minimizing the
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77 result. The fragments should both contain 2-byte samples. Return a
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78 tuple \code{(\var{offset}, \var{factor})} where \var{offset} is the
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79 (integer) offset into \var{fragment} where the optimal match started
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80 and \var{factor} is the (floating-point) factor as per
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81 \code{findfactor}.
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82 \end{funcdesc}
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83
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84 \begin{funcdesc}{findmax}{fragment\, length}
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85 Search \var{fragment} for a slice of length \var{length} samples (not
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86 bytes!)\ with maximum energy, i.e., return \var{i} for which
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87 \code{rms(fragment[i*2:(i+length)*2])} is maximal. The fragments
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88 should both contain 2-byte samples.
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89
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90 The routine takes time proportional to \code{len(fragment)}.
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91 \end{funcdesc}
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92
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93 \begin{funcdesc}{getsample}{fragment\, width\, index}
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94 Return the value of sample \var{index} from the fragment.
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95 \end{funcdesc}
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96
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97 \begin{funcdesc}{lin2lin}{fragment\, width\, newwidth}
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98 Convert samples between 1-, 2- and 4-byte formats.
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99 \end{funcdesc}
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100
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101 \begin{funcdesc}{lin2adpcm}{fragment\, width\, state}
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102 Convert samples to 4 bit Intel/DVI ADPCM encoding. ADPCM coding is an
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103 adaptive coding scheme, whereby each 4 bit number is the difference
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104 between one sample and the next, divided by a (varying) step. The
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105 Intel/DVI ADPCM algorithm has been selected for use by the IMA, so it
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106 may well become a standard.
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107
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108 \code{State} is a tuple containing the state of the coder. The coder
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109 returns a tuple \code{(\var{adpcmfrag}, \var{newstate})}, and the
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110 \var{newstate} should be passed to the next call of lin2adpcm. In the
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111 initial call \code{None} can be passed as the state. \var{adpcmfrag}
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112 is the ADPCM coded fragment packed 2 4-bit values per byte.
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113 \end{funcdesc}
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114
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115 \begin{funcdesc}{lin2adpcm3}{fragment\, width\, state}
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116 This is an alternative ADPCM coder that uses only 3 bits per sample.
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117 It is not compatible with the Intel/DVI ADPCM coder and its output is
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118 not packed (due to laziness on the side of the author). Its use is
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119 discouraged.
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120 \end{funcdesc}
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121
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122 \begin{funcdesc}{lin2ulaw}{fragment\, width}
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123 Convert samples in the audio fragment to U-LAW encoding and return
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124 this as a Python string. U-LAW is an audio encoding format whereby
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125 you get a dynamic range of about 14 bits using only 8 bit samples. It
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126 is used by the Sun audio hardware, among others.
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127 \end{funcdesc}
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128
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129 \begin{funcdesc}{minmax}{fragment\, width}
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130 Return a tuple consisting of the minimum and maximum values of all
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131 samples in the sound fragment.
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132 \end{funcdesc}
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133
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134 \begin{funcdesc}{max}{fragment\, width}
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135 Return the maximum of the {\em absolute value} of all samples in a
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136 fragment.
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137 \end{funcdesc}
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139 \begin{funcdesc}{maxpp}{fragment\, width}
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140 Return the maximum peak-peak value in the sound fragment.
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141 \end{funcdesc}
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142
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143 \begin{funcdesc}{mul}{fragment\, width\, factor}
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144 Return a fragment that has all samples in the original framgent
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145 multiplied by the floating-point value \var{factor}. Overflow is
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146 silently ignored.
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147 \end{funcdesc}
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148
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149 \begin{funcdesc}{ratecv}{fragment\, width\, nchannels\, inrate\, outrate\, state\optional{\, weightA\, weightB}}
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150 Convert the frame rate of the input fragment.
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151
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152 \code{State} is a tuple containing the state of the converter. The
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153 converter returns a tupl \code{(\var{newfragment}, \var{newstate})},
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154 and \var{newstate} should be passed to the next call of ratecv.
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155
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156 The \code{weightA} and \code{weightB} arguments are parameters for a
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157 simple digital filter and default to 1 and 0 respectively.
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158 \end{funcdesc}
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159
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160 \begin{funcdesc}{reverse}{fragment\, width}
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161 Reverse the samples in a fragment and returns the modified fragment.
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162 \end{funcdesc}
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163
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164 \begin{funcdesc}{rms}{fragment\, width}
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165 Return the root-mean-square of the fragment, i.e.
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166 \iftexi
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167 the square root of the quotient of the sum of all squared sample value,
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168 divided by the sumber of samples.
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169 \else
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170 % in eqn: sqrt { sum S sub i sup 2 over n }
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171 \begin{displaymath}
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172 \catcode`_=8
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173 \sqrt{\frac{\sum{{S_{i}}^{2}}}{n}}
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174 \end{displaymath}
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175 \fi
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176 This is a measure of the power in an audio signal.
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177 \end{funcdesc}
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178
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179 \begin{funcdesc}{tomono}{fragment\, width\, lfactor\, rfactor}
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180 Convert a stereo fragment to a mono fragment. The left channel is
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181 multiplied by \var{lfactor} and the right channel by \var{rfactor}
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182 before adding the two channels to give a mono signal.
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183 \end{funcdesc}
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184
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185 \begin{funcdesc}{tostereo}{fragment\, width\, lfactor\, rfactor}
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186 Generate a stereo fragment from a mono fragment. Each pair of samples
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187 in the stereo fragment are computed from the mono sample, whereby left
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188 channel samples are multiplied by \var{lfactor} and right channel
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189 samples by \var{rfactor}.
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190 \end{funcdesc}
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191
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192 \begin{funcdesc}{ulaw2lin}{fragment\, width}
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193 Convert sound fragments in ULAW encoding to linearly encoded sound
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194 fragments. ULAW encoding always uses 8 bits samples, so \var{width}
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195 refers only to the sample width of the output fragment here.
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196 \end{funcdesc}
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197
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198 Note that operations such as \code{mul} or \code{max} make no
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199 distinction between mono and stereo fragments, i.e.\ all samples are
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200 treated equal. If this is a problem the stereo fragment should be split
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201 into two mono fragments first and recombined later. Here is an example
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202 of how to do that:
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203 \bcode\begin{verbatim}
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204 def mul_stereo(sample, width, lfactor, rfactor):
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205 lsample = audioop.tomono(sample, width, 1, 0)
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206 rsample = audioop.tomono(sample, width, 0, 1)
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207 lsample = audioop.mul(sample, width, lfactor)
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208 rsample = audioop.mul(sample, width, rfactor)
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209 lsample = audioop.tostereo(lsample, width, 1, 0)
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210 rsample = audioop.tostereo(rsample, width, 0, 1)
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211 return audioop.add(lsample, rsample, width)
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212 \end{verbatim}\ecode
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213
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214 If you use the ADPCM coder to build network packets and you want your
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215 protocol to be stateless (i.e.\ to be able to tolerate packet loss)
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216 you should not only transmit the data but also the state. Note that
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217 you should send the \var{initial} state (the one you passed to
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218 \code{lin2adpcm}) along to the decoder, not the final state (as returned by
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219 the coder). If you want to use \code{struct} to store the state in
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220 binary you can code the first element (the predicted value) in 16 bits
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221 and the second (the delta index) in 8.
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222
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223 The ADPCM coders have never been tried against other ADPCM coders,
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224 only against themselves. It could well be that I misinterpreted the
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225 standards in which case they will not be interoperable with the
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226 respective standards.
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227
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228 The \code{find...} routines might look a bit funny at first sight.
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229 They are primarily meant to do echo cancellation. A reasonably
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230 fast way to do this is to pick the most energetic piece of the output
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231 sample, locate that in the input sample and subtract the whole output
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232 sample from the input sample:
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233 \bcode\begin{verbatim}
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234 def echocancel(outputdata, inputdata):
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235 pos = audioop.findmax(outputdata, 800) # one tenth second
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236 out_test = outputdata[pos*2:]
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237 in_test = inputdata[pos*2:]
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238 ipos, factor = audioop.findfit(in_test, out_test)
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239 # Optional (for better cancellation):
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240 # factor = audioop.findfactor(in_test[ipos*2:ipos*2+len(out_test)],
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241 # out_test)
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242 prefill = '\0'*(pos+ipos)*2
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243 postfill = '\0'*(len(inputdata)-len(prefill)-len(outputdata))
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244 outputdata = prefill + audioop.mul(outputdata,2,-factor) + postfill
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245 return audioop.add(inputdata, outputdata, 2)
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246 \end{verbatim}\ecode