Secret Rabbit Code (aka libsamplerate)
Author: Erik de Castro Lopo (erikd AT mega-nerd DOT com)
Description: The Secret Rabbit Code Home Page
Keywords: libsamplerate sound resample audio dsp Linux

| # Miscellaneous API Documentation ### Error Reporting Most of the API functions either return an integer error (ie src_simple and src_process) or return an integer error value via an int pointer parameter (src_new). These integer error values can be converted into a human readable text strings by calling the function:
          const char* src_strerror (int error) ;
which will return an error string for valid error numbers, the string "No Error" for an error value of zero or a NULL pointer if no error message has been defined for that error value. ### Converters Secret Rabbit Code has a number of different converters which can be selected using the converter_type parameter when calling src_simple or src_new. Currently, the five converters available are:
          enum
          {
              SRC_SINC_BEST_QUALITY       = 0,
              SRC_SINC_MEDIUM_QUALITY     = 1,
              SRC_SINC_FASTEST            = 2,
              SRC_ZERO_ORDER_HOLD         = 3,
              SRC_LINEAR                  = 4
          } ;
As new converters are added, they will given a number corresponding to the next inetger. The details of these converters are as follows: SRC_SINC_BEST_QUALITY - This is a bandlimited interpolator derived from the mathematical sinc function and this is the highest quality sinc based converter, providing a worst case Signal-to-Noise Ratio (SNR) of 97 decibels (dB) at a bandwidth of 97%. All three SRC_SINC_* converters are based on the techniques of Julius O. Smith although this code was developed independantly. SRC_SINC_MEDIUM_QUALITY - This is another bandlimited interpolator much like the previous one. It has an SNR of 97dB and a bandwidth of 90%. The speed of the conversion is much faster than the previous one. SRC_SINC_FASTEST - This is the fastest bandlimited interpolator and has an SNR of 97dB and a bandwidth of 80%. SRC_ZERO_ORDER_HOLD - A Zero Order Hold converter (interpolated value is equal to the last value). The quality is poor but the conversion speed is blindlingly fast. SRC_LINEAR - A linear converter. Again the quality is poor, but the conversion speed is blindingly fast. There are two functions that give either a (text string) name or description for each converter:
          const char *src_get_name (int converter_type) ;
          const char *src_get_description (int converter_type) ;
The name will typically be a short string for use in a dialog box, while the description string is longer. Both of these functions return a NULL pointer if there is no converter for the given converter_type value. Since the converters have consecutive converter_type values, the caller is easily able to figure out the number of converters at run time. This enables a binary dynamically linked against an old version of the library to know about converters from later versions of the library as they become available. ### SRC_DATA Both the simple and the full featured versions of the API use the SRC_DATA struct to pass audio and control data into the sample rate converter. This struct is defined as:
          typedef struct
          {   float  *data_in, *data_out ;

              long   input_frames, output_frames ;
              long   input_frames_used, output_frames_gen ;

              int    end_of_input ;

              double src_ratio ;
          } SRC_DATA ;
The data_in pointer is used to pass audio data into the converter while the data_out pointer supplies the converter with an array to hold the converter's output. For a converter which has been configured for mulitchannel operation, these pointers need to point to a single array of interleaved data. The input_frames and output_frames fields supply the converter with the lengths of the arrays (in frames) pointed to by the data_in and data_out pointers respectively. For monophinc data, these values would indicate the length of the arrays while for multi channel data these values would be equal to the the length of the array divided by the number of channels. The end_of_input field is only used when the sample rate converter is used by calling the src_process function. In this case it should be set to zero if more buffers are to be passed to the converter and 1 if the current buffer is the last. Finally, the src_ratio field specifies the conversion ratio defined as the input sample rate divided by the output sample rate. For a connected set of buffers, this value can be varies on each call to src_process resulting in a time varying sample rate conversion process. For time varying sample rate conversions, the ratio will be linearly interpolated between the src_ratio value of the previous call to src_process and the value for the current call. The input_frames_used and output_frames_gen fields are set by the converter to inform the caller of the number of frames consumed from the data_in array and the number of frames generated in the data_out array respectively. These values are for the current call to src_process only. ### Auxillary Functions There are four auxillary functions for converting arrays of float data to and from short or int data. These functions are defined as:
        void src_short_to_float_array (const short *in, float *out, int len) ;
        void src_float_to_short_array (const float *in, short *out, int len) ;
        void src_int_to_float_array (const int *in, float *out, int len) ;
        void src_float_to_int_array (const float *in, int *out, int len) ;
The float data is assumed to be in the range [-1.0, 1.0] and it is automatically scaled on the conversion to and from float. On the float to short/int conversion path, any data values which would overflow the range of short/int data are clipped.