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Frequently Asked Questions
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cannam@85: Q1 : Is it normal for the output of libsamplerate to be louder
cannam@85: than its input?
cannam@85: Q2 : On Unix/Linux/MacOSX, what is the best way of detecting
cannam@85: the presence and location of libsamplerate and its header file using
cannam@85: autoconf?
cannam@85: Q3 : If I upsample and downsample to the original rate, for
cannam@85: example 44.1->96->44.1, do I get an identical signal as the one before the
cannam@85: up/down resampling?
cannam@85: Q4 : If I ran src_simple (libsamplerate) on small chunks (160
cannam@85: frames) would that sound bad?
cannam@85: Q5 : I'm using libsamplerate but the high quality settings
cannam@85: sound worse than the SRC_LINEAR converter. Why?
cannam@85: Q6 : I'm use the SRC_SINC_* converters and up-sampling by a ratio of
cannam@85: 2. I reset the converter and put in 1000 samples and I expect to get 2000
cannam@85: samples out, but I'm getting less than that. Why?
cannam@85: Q7 : I have input and output sample rates that are integer
cannam@85: values, but the API wants me to divide one by the other and put the result
cannam@85: in a floating point number. Won't this case problems for long running
cannam@85: conversions?
cannam@85:
cannam@85:
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cannam@85:
cannam@85:
Q1 : Is it normal for the output of libsamplerate to be louder
cannam@85: than its input?
cannam@85:
cannam@85: The output of libsamplerate will be roughly the same volume as the input.
cannam@85: However, even if the input is strictly in the range (-1.0, 1.0), it is still
cannam@85: possible for the output to contain peak values outside this range.
cannam@85:
cannam@85:
cannam@85: Consider four consecutive samples of [0.5 0.999 0.999 0.5].
cannam@85: If we are up sampling by a factor of two we need to insert samples between
cannam@85: each of the existing samples.
cannam@85: Its pretty obvious then, that the sample between the two 0.999 values should
cannam@85: and will be bigger than 0.999.
cannam@85:
cannam@85:
cannam@85: This means that anyone using libsamplerate should normalize its output before
cannam@85: doing things like saving the audio to a 16 bit WAV file.
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cannam@85:
cannam@85:
Q2 : On Unix/Linux/MacOSX, what is the best way of detecting
cannam@85: the presence and location of libsamplerate and its header file using
cannam@85: autoconf?
cannam@85:
cannam@85:
cannam@85: libsamplerate uses the pkg-config (man pkg-config) method of registering itself
cannam@85: with the host system.
cannam@85: The best way of detecting its presence is using something like this in configure.ac
cannam@85: (or configure.in):
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cannam@85:
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cannam@85: PKG_CHECK_MODULES(SAMPLERATE, samplerate >= 0.1.3,
cannam@85: ac_cv_samplerate=1, ac_cv_samplerate=0)
cannam@85:
cannam@85: AC_DEFINE_UNQUOTED([HAVE_SAMPLERATE],${ac_cv_samplerate},
cannam@85: [Set to 1 if you have libsamplerate.])
cannam@85:
cannam@85: AC_SUBST(SAMPLERATE_CFLAGS)
cannam@85: AC_SUBST(SAMPLERATE_LIBS)
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cannam@85:
cannam@85: This will automatically set the SAMPLERATE_CFLAGS and SAMPLERATE_LIBS
cannam@85: variables which can be used in Makefile.am or Makefile.in like this:
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cannam@85: SAMPLERATE_CFLAGS = @SAMPLERATE_CFLAGS@
cannam@85: SAMPLERATE_LIBS = @SAMPLERATE_LIBS@
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cannam@85:
cannam@85:
cannam@85: If you install libsamplerate from source, you will probably need to set the
cannam@85: PKG_CONFIG_PATH environment variable's suggested at the end of the
cannam@85: libsamplerate configure process. For instance on my system I get this:
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cannam@85: -=-=-=-=-=-=-=-=-=-= Configuration Complete =-=-=-=-=-=-=-=-=-=-=-
cannam@85:
cannam@85: Configuration summary :
cannam@85:
cannam@85: Version : ..................... 0.1.3
cannam@85: Enable debugging : ............ no
cannam@85:
cannam@85: Tools :
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cannam@85: Compiler is GCC : ............. yes
cannam@85: GCC major version : ........... 3
cannam@85:
cannam@85: Extra tools required for testing and examples :
cannam@85:
cannam@85: Have FFTW : ................... yes
cannam@85: Have libsndfile : ............. yes
cannam@85: Have libefence : .............. no
cannam@85:
cannam@85: Installation directories :
cannam@85:
cannam@85: Library directory : ........... /usr/local/lib
cannam@85: Program directory : ........... /usr/local/bin
cannam@85: Pkgconfig directory : ......... /usr/local/lib/pkgconfig
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Q3 : If I upsample and downsample to the original rate, for
cannam@85: example 44.1->96->44.1, do I get an identical signal as the one before the
cannam@85: up/down resampling?
cannam@85:
cannam@85: The short answer is that for the general case, no, you don't.
cannam@85: The long answer is that for some signals, with some converters, you will
cannam@85: get very, very close.
cannam@85:
cannam@85:
cannam@85: In order to resample correctly (ie using the SRC_SINC_* converters),
cannam@85: filtering needs to be applied, regardless of whether its upsampling or
cannam@85: downsampling.
cannam@85: This filter needs to attenuate all frequencies above 0.5 times the minimum of
cannam@85: the source and destination sample rate (call this fshmin).
cannam@85: Since the filter needed to achieve full attenuation at this point, it has to
cannam@85: start rolling off a some frequency below this point.
cannam@85: It is this rolloff of the very highest frequencies which causes some of the
cannam@85: loss.
cannam@85:
cannam@85:
cannam@85: The other factor is that the filter itself can introduce transient artifacts
cannam@85: which causes the output to be different to the input.
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cannam@85:
cannam@85:
Q4 : If I ran src_simple on small chunks (say 160 frames) would that
cannam@85: sound bad?
cannam@85:
cannam@85: Well if you are after odd sound effects, it might sound OK.
cannam@85: If you are after high quality sample rate conversion you will be disappointed.
cannam@85:
cannam@85:
cannam@85: The src_simple() was designed to provide a simple to use interface for people
cannam@85: who wanted to do sample rate conversion on say, a whole file all at once.
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cannam@85:
cannam@85:
Q5 : I'm using libsamplerate but the high quality settings
cannam@85: sound worse than the SRC_LINEAR converter. Why?
cannam@85:
cannam@85: There are two possible problems.
cannam@85: Firstly, if you are using the src_simple() function on successive blocks
cannam@85: of a stream of samples, you will get bad results. The src_simple() function
cannam@85: is designed for use on a whole sound file, all at once, not on contiguous
cannam@85: segments of the same sound file.
cannam@85: To fix the problem, you need to move to the src_process() API or the callback
cannam@85: based API.
cannam@85:
cannam@85:
cannam@85: If you are already using the src_process() API or the callback based API and
cannam@85: the high quality settings sound worse than SRC_LINEAR, then you have other
cannam@85: problems.
cannam@85: Read on for more debugging hints.
cannam@85:
cannam@85:
cannam@85: All of the higher quality converters need to keep state while doing conversions
cannam@85: on segments of a large chunk of audio.
cannam@85: This state information is kept inside the private data pointed to by the
cannam@85: SRC_STATE pointer returned by the src_new() function.
cannam@85: This means, that when you want to start doing sample rate conversion on a
cannam@85: stream of data, you should call src_new() to get a new SRC_STATE pointer
cannam@85: (or alternatively, call src_reset() on an existing SRC_STATE pointer).
cannam@85: You should then pass this SRC_STATE pointer to the src_process() function
cannam@85: with each new block of audio data.
cannam@85: When you have completed the conversion, you can then call src_delete() on
cannam@85: the SRC_STATE pointer.
cannam@85:
cannam@85:
cannam@85: If you are doing all of the above correctly, you need to examine your usage
cannam@85: of the values passed to src_process() in the
cannam@85: SRC_DATA
cannam@85: struct.
cannam@85: Specifically:
cannam@85:
cannam@85:
cannam@85: - Check that input_frames and output_frames fields are being set in
cannam@85: terms of frames (number of sample values times channels) instead
cannam@85: of just the number of samples.
cannam@85:
- Check that you are using the return values input_frames_used and
cannam@85: output_frames_gen to update your source and destination pointers
cannam@85: correctly.
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- Check that you are updating the data_in and data_out pointers
cannam@85: correctly for each successive call.
cannam@85:
cannam@85:
cannam@85: While doing the above, it is probably useful to compare what you are doing to
cannam@85: what is done in the example programs in the examples/ directory of the source
cannam@85: code tarball.
cannam@85:
cannam@85:
cannam@85: If you have done all of the above and are still having problems then its
cannam@85: probably time to email the author with the smallest chunk of code that
cannam@85: adequately demonstrates your problem.
cannam@85: This chunk should not need to be any more than 100 lines of code.
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Q6 : I'm use the SRC_SINC_* converters and up-sampling by a ratio of
cannam@85: 2. I reset the converter and put in 1000 samples and I expect to get 2000
cannam@85: samples out, but I'm getting less than that. Why?
cannam@85:
cannam@85: The short answer is that there is a transport delay inside the converter itself.
cannam@85: Long answer follows.
cannam@85:
cannam@85:
cannam@85: By way of example, the first time you call src_process() you might only get 1900
cannam@85: samples out.
cannam@85: However, after that first call all subsequent calls will probably get you about
cannam@85: 2000 samples out for every 1000 samples you put in.
cannam@85:
cannam@85:
cannam@85: The main problems people have with this transport delay is that they need to read
cannam@85: out an exact number of samples and the transport delay scews this up.
cannam@85: The best way to overcome this problem is to always supply more samples on the
cannam@85: input than is actually needed to create the required number of output samples.
cannam@85: With reference to the example above, if you always supply 1500 samples at the
cannam@85: input, you will always get 2000 samples at the output.
cannam@85: You will always need to keep track of the number of input frames used on each
cannam@85: call to src_process() and deal with these values appropriately.
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cannam@85:
cannam@85:
Q7 : I have input and output sample rates that are integer
cannam@85: values, but the API wants me to divide one by the other and put the result
cannam@85: in a floating point number. Won't this case problems for long running
cannam@85: conversions?
cannam@85:
cannam@85: The short answer is no, the precision of the ratio is many orders of magnitude
cannam@85: more than is really needed.
cannam@85:
cannam@85:
cannam@85: For the long answer, lets do come calculations.
cannam@85: Firstly, the src_ratio field is double precision floating point number
cannam@85: which has
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cannam@85: 53 bits of precision.
cannam@85:
cannam@85:
cannam@85: That means that the maximum error in your ratio converted to a double is one
cannam@85: bit in 2^53 which means the the double float value would be wrong by one sample
cannam@85: after 9007199254740992 samples have passed or wrong by more than half a sample
cannam@85: wrong after half that many (4503599627370496 samples) have passed.
cannam@85:
cannam@85:
cannam@85: Now if for example our output sample rate is 96kHz then
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cannam@85:
cannam@85: 4503599627370496 samples at 96kHz is 46912496118 seconds
cannam@85: 46912496118 seconds is 781874935 minutes
cannam@85: 781874935 minutes is 13031248 hours
cannam@85: 13031248 hours is 542968 days
cannam@85: 542968 days is 1486 years
cannam@85:
cannam@85:
cannam@85: So, after 1486 years, the input will be wrong by more than half of one sampling
cannam@85: period.
cannam@85:
cannam@85:
cannam@85: All this assumes that the crystal oscillators uses to sample the audio stream
cannam@85: is perfect.
cannam@85: This is not the case.
cannam@85: According to
cannam@85:
cannam@85: this web site,
cannam@85: the accuracy of standard crystal oscillators (XO, TCXO, OCXO) is at best
cannam@85: 1 in 100 million.
cannam@85: The src_ratio is therefore 45035996 times more accurate than the
cannam@85: crystal clock source used to sample the original audio signal and any potential
cannam@85: problem with the src_ratio being a floating point number will be
cannam@85: completely swamped by sampling inaccuracies.
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