ref: bb11b7ac761dfa5a72d1c16743aecab697301248
parent: d1c93aa5ef421077ac4516d14179b63ff05f1fbf
author: Erik de Castro Lopo <erikd@miles>
date: Sun May 29 22:02:11 EDT 2005
Update FAQ with hints on debugging choppy output audio.
--- a/doc/faq.html
+++ b/doc/faq.html
@@ -69,6 +69,8 @@
up/down resampling?</A><BR><BR>
<A HREF="#Q004">Q4 : If I ran src_simple (libsamplerate) on small chunks (160
frames) would that sound bad?</A><BR><BR>
+<A HREF="#Q005">Q5 : I'm using libsamplerate but the high quality settings
+ sound worse than the SRC_LINEAR converter. Why?</A><BR><BR>
</P>
<HR>
<!-- ========================================================================= -->
@@ -88,7 +90,7 @@
and will be bigger than 0.999.
</P>
<P>
-This means that anyone using libsamplerate should normalise its output before
+This means that anyone using libsamplerate should normalize its output before
doing things like saving the audio to a 16 bit WAV file.
</P>
@@ -100,7 +102,7 @@
autoconf?</b></h2>
<p>
-libsamplrate uses the pkg-config (man pkg-config) method of registering itself
+libsamplerate uses the pkg-config (man pkg-config) method of registering itself
with the host system.
The best way of detecting its presence is using something like this in configure.ac
(or configure.in):
@@ -171,9 +173,9 @@
In order to resample correctly (ie using the <B>SRC_SINC_*</B> converters),
filtering needs to be applied, regardless of whether its upsampling or
downsampling.
-This filter needs to attenuate all frequencis above 0.5 times the minimum of
-the source and desination sample rate (call this fshmin).
-Since the filter needed to achieve full attemuation at this point, it has to
+This filter needs to attenuate all frequencies above 0.5 times the minimum of
+the source and destination sample rate (call this fshmin).
+Since the filter needed to achieve full attenuation at this point, it has to
start rolling off a some frequency below this point.
It is this rolloff of the very highest frequencies which causes some of the
loss.
@@ -194,6 +196,67 @@
<P>
The src_simple() was designed to provide a simple to use interface for people
who wanted to do sample rate conversion on say, a whole file all at once.
+</P>
+
+<!-- ========================================================================= -->
+<A NAME="Q005"></A>
+<H2><BR><B>Q5 : I'm using libsamplerate but the high quality settings
+ sound worse than the SRC_LINEAR converter. Why?</B></H2>
+<P>
+There are two possible problems.
+Firstly, if you are using the src_simple() function on successive blocks
+of a stream of samples, you will get bad results. The src_simple() function
+is designed for use on a whole sound file, all at once, not on contiguous
+segments of the same sound file.
+To fix the problem, you need to move to the src_process() API or the callback
+based API.
+</P>
+<P>
+If you are already using the src_process() API or the callback based API and
+the high quality settings sound worse than SRC_LINEAR, then you have other
+problems.
+Read on for more debugging hints.
+</P>
+<P>
+All of the higher quality converters need to keep state while doing conversions
+on segments of a large chunk of audio.
+This state information is kept inside the private data pointed to by the
+SRC_STATE pointer returned by the src_new() function.
+This means, that when you want to start doing sample rate conversion on a
+stream of data, you should call src_new() to get a new SRC_STATE pointer
+(or alternatively, call src_reset() on an existing SRC_STATE pointer).
+You should then pass this SRC_STATE pointer to the src_process() function
+with each new block of audio data.
+When you have completed the conversion, you can then call src_delete() on
+the SRC_STATE pointer.
+</P>
+<P>
+If you are doing all of the above correctly, you need to examine your usage
+of the values passed to src_process() in the
+ <A HREF="api_misc.html#SRC_DATA">SRC_DATA</A>
+struct.
+Specifically:
+</P>
+<UL>
+ <LI> Check that input_frames and output_frames fields are being set in
+ terms of frames (number of sample values times channels) instead
+ of just the number of samples.
+ <LI> Check that you are using the return values input_frames_used and
+ output_frames_gen to update your source and destination pointers
+ correctly.
+ <LI> Check that you are updating the data_in and data_out pointers
+ correctly for each successive call.
+</UL>
+<P>
+While doing the above, it is probably useful to compare what you are doing to
+what is done in the example programs in the examples/ directory of the source
+code tarball.
+</P>
+<P>
+If you have done all of the above and are still having problems then its
+probably time to email the author with the smallest chunk of code that
+adequately demonstrates your problem.
+This chunk should not need to be any more than 100 lines of code.
</P>
<!-- <A HREF="mailto:aldel@mega-nerd.com">For the spam bots</A> -->