fix formatting, etc.

README.md

@@ -4,7 +4,7 @@ libcsdr
 *libcsdr* is a set of simple DSP routines for Software Defined Radio.  
 It is mostly useful for AM/FM/SSB demodulation and spectrum display.  
 Feel free to use it in your projects.  
-Most of the code is available under the permissive BSD license, with some optional parts under GPL. For additional details, see [Licensing].
+Most of the code is available under the permissive BSD license, with some optional parts under GPL. For additional details, see <a href="licensing">licensing</a>.
 
 - The package comes with a command-line tool `csdr`, which lets you build DSP processing chains by shell pipes.
 - The code of *libcsdr* was intended to be easy to follow.
@@ -48,7 +48,8 @@ Usage by example
 - We want to listen to one radio station, but input signal contains multiple stations, and its bandwidth is too large for sending it directly to the FM demodulator.
 - We shift the signal to the center frequency of the station we want to receive: `-0.17*2400000*0.5 = -204000`, so basically we will listen to the radio station centered at 89504000 Hz.
 - We decimate the signal by a factor of 10. The rolloff for the FIR filter used for decimation will be 10% of total bandwidth (as of parameter 0.05 is 10% of 0.5). Hamming window will be used for windowed FIR filter design.
-- Sample rates look like this:
+
+Sample rates look like this:
 
 
 	             2.4 Msps                     240 ksps                                  48 ksps
@@ -134,33 +135,33 @@ You can use these commands on complex streams, too, as they are only interleaved
 Regarding `csdr`, the first command-line parameter is the name of a function, others are the parameters for the given function. Compulsory parameters are noted as `<parameter>`, optional parameters are noted as `[parameter]`.
 Optional parameters have safe defaults, for more info look at the code.
 
-##### `realpart_cf`
+	realpart_cf
 
 It takes the real part of the complex signal, and throws away the imaginary part.
 
-##### `clipdetect_ff`
+	clipdetect_ff
 
 It clones the signal (the input and the output is the same), but it prints a warning on `stderr` if any sample value is out of the -1.0 ... 1.0 range.
 
-##### `limit_ff [max_amplitude]`
+	limit_ff [max_amplitude]
 
 The input signal amplitude will not be let out of the `-max_amplitude ... max_amplitude` range.
 
-##### `gain_ff <gain>`
+	gain_ff <gain>
 
 It multiplies all samples by `gain`.
 
-##### `clone`
+	clone
 
 It copies the input to the output.
 
-##### `yes_f <to_repeat> [buf_times]`
+	yes_f <to_repeat> [buf_times]
 
 It outputs continously the `to_repeat` float number. 
 If `buf_times` is not given, it never stops.
 Else, after outputing `buf_times` number of buffers (the size of which is stated in the `BUFSIZE` macro), it exits.
 
-##### `shift_math_cc <rate>`
+	shift_math_cc <rate>
 
 It shifts the complex spectrum by `rate`.
 `rate` is a floating point number between -0.5 and 0.5. 
@@ -168,56 +169,56 @@ It shifts the complex spectrum by `rate`.
 
 Internally, a sine and cosine wave is generated to perform this function, and this function uses `math.h` for this purpose, which is quite accurate, but not always very fast.
 
-#####  `shift_addition_cc <rate>`
+	shift_addition_cc <rate>
 
 Operation is the same as with `shift_math_cc`.
 
 Internally, this function uses trigonometric addition formulas to generate sine and cosine, which is a bit faster. (About 4 times on the machine I have tested it on.)
 
-##### `shift_addition_cc_test`
+	shift_addition_cc_test
 
 This function was used to test the accuracy of the method above.
 
-##### `dcblock_ff`
+	dcblock_ff
 
 This is a DC blocking IIR filter.
 
-##### `fastdcblock_ff`
+	fastdcblock_ff
 
 This is a DC blocker that works based on the average of the buffer.
 
-##### `fmdemod_atan_cf`
+	fmdemod_atan_cf
 
 It is an FM demodulator that internally uses the `atan` function in `math.h`, so it is not so fast.
 
-##### `fmdemod_quadri_cf`
+	fmdemod_quadri_cf
 
 It is an FM demodulator that is based on the quadri-correlator method, and it can be effectively auto-vectorized, so it should be faster.
 
-##### `fmdemod_quadri_novect_cf`
+	fmdemod_quadri_novect_cf
 
 It has more easily understandable code than the previous one, but can't be auto-vectorized.
 
-##### `deemphasis_wfm_ff <sample_rate> <tau>`
+	deemphasis_wfm_ff <sample_rate> <tau>
 
 It does de-emphasis with the given RC time constant `tau`.
 Different parts of the world use different pre-emphasis filters for FM broadcasting.
 In Europe, `tau` should be chosen as `50e-6`, and in the USA, `tau` should be `75e-6`.
 
-##### `deemphasis_nfm_ff <one_of_the_predefined_sample_rates>`
+	deemphasis_nfm_ff <one_of_the_predefined_sample_rates>
 
 It does de-emphasis on narrow-band FM for communication equipment (e.g. two-way radios).
 It uses fixed filters so it works only on predefined sample rates, for the actual list of them run: `cat libcsdr.c | grep DNFMFF_ADD_ARRAY`
 
-##### `amdemod_cf`
+	amdemod_cf
 
 It is an AM demodulator that uses `sqrt`. On some architectures `sqrt` can be directly calculated by dedicated CPU instructions, but on others it may be slower. 
 
-##### `amdemod_estimator_cf`
+amdemod_estimator_cf
 
 It is an AM demodulator that uses an estimation method that is faster but less accurate than `amdemod_cf`.
 
-##### `firdes_lowpass_f <cutoff_rate> <length> [window [--octave]]`
+	firdes_lowpass_f <cutoff_rate> <length> [window [--octave]]
 
 Low-pass FIR filter design function to output real taps, with a `cutoff_rate` proportional to the sampling frequency, using the windowed sinc filter design method.
 `cutoff_rate` can be between 0 and 0.5.
@@ -231,40 +232,40 @@ Some functions (below) require the `transition_bw` to be given instead of filter
 
 The `--octave` parameter lets you directly view the filter response in `octave`. For more information, look at the [Usage by example] section.
 
-##### `firdes_bandpass_c <low_cut> <high_cut> <length> [window [--octave]]`
+	firdes_bandpass_c <low_cut> <high_cut> <length> [window [--octave]]
 
 Band-pass FIR filter design function to output complex taps.
 `low_cut` and ` high_cut` both may be between -0.5 and 0.5, and are also proportional to the sampling frequency.
 
 Other parameters were explained above at `firdes_lowpass_f`.
 
-##### `fir_decimate_cc <decimation_factor> [transition_bw [window]]`
+	fir_decimate_cc <decimation_factor> [transition_bw [window]]
 
 It is a decimator that keeps one sample out of `decimation_factor` samples. 
 To avoid aliasing, it runs a filter on the signal and removes spectral components above `0.5 × nyquist_frequency × decimation_factor`.
 
 `transition_bw` and `window` are the parameters of the filter.
 
-##### `rational_resampler_ff <interpolation> <decimation> [transition_bw [window]]`
+	rational_resampler_ff <interpolation> <decimation> [transition_bw [window]]
 
 It is a resampler that takes integer values of `interpolation` and `decimation`.
 The output sample rate will be `interpolation / decimation × input_sample_rate`.
 
 `transition_bw` and `window` are the parameters of the filter.
 
-##### `fractional_decimator_ff <decimation_rate> [transition_bw [window]]`
+	fractional_decimator_ff <decimation_rate> [transition_bw [window]]
 
 It can decimate by a floating point ratio.
 
 `transition_bw` and `window` are the parameters of the filter.
 
-#### `bandpass_fir_fft_cc <low_cut> <high_cut> <transition_bw> [window]`
+	bandpass_fir_fft_cc <low_cut> <high_cut> <transition_bw> [window]
 
 It performs a bandpass FIR filter on complex samples, using FFT and the overlap-add method.
 
 Parameters are described under `firdes_bandpass_c` and `firdes_lowpass_f`.
 
-##### `agc_ff [hang_time [reference [attack_rate [decay_rate [max_gain [attack_wait [filter_alpha]]]]]]]`
+	agc_ff [hang_time [reference [attack_rate [decay_rate [max_gain [attack_wait [filter_alpha]]]]]]]
 
 It is an automatic gain control function. 
 
@@ -278,11 +279,11 @@ It is an automatic gain control function.
 
 Its default parameters work best for an audio signal sampled at 48000 Hz.
 
-##### `fastagc_ff [block_size [reference]]`
+	fastagc_ff [block_size [reference]]
 
 It is a faster AGC that linearly changes the gain, taking the highest amplitude peak in the buffer into consideration. Its output will never exceed `-reference ... reference`.
 
-##### `fft_cc <fft_size> <out_of_every_n_samples> [window [--octave] [--benchmark]]`
+	fft_cc <fft_size> <out_of_every_n_samples> [window [--octave] [--benchmark]]
 
 It performs an FFT on the first `fft_size` samples out of `out_of_every_n_samples`, thus skipping `out_of_every_n_samples - fft_size` samples in the input. 
 
@@ -290,12 +291,12 @@ It can draw the spectrum by using `--octave`, for more information, look at the
 
 FFTW can be faster if we let it optimalize a while before starting the first transform, hence the `--benchmark` switch.
 
-##### `fft_benchmark <fft_size> <fft_cycles> [--benchmark]`
+	fft_benchmark <fft_size> <fft_cycles> [--benchmark]
 
 It measures the time taken to process `fft_cycles` transforms of `fft_size`.
 It lets FFTW optimalize if used with the `--benchmark` switch.
 
-##### `lowpower_cf [add_db]`
+	lowpower_cf [add_db]
 
 Calculates `10*log10(i^2+q^2)+add_db` for the input complex samples. It is useful for drawing power spectrum graphs.
 
@@ -303,7 +304,7 @@ Calculates `10*log10(i^2+q^2)+add_db` for the input complex samples. It is usefu
 
 Some parameters can be changed while the `csdr` process is running. To achieve this, some `csdr` functions have special parameters. You have to supply a fifo previously created by the `mkfifo` command. Processing will only start after the first control command has been received by `csdr` over the FIFO.
 
-##### `shift_addition_cc --fifo <fifo_path>`
+	shift_addition_cc --fifo <fifo_path>
 
 By writing to the given FIFO file with the syntax below, you can control the shift rate:
 
@@ -313,7 +314,7 @@ E.g. you can send `-0.3\n`
 
 Processing will only start after the first control command has been received by `csdr` over the FIFO.
 
-##### `bandpass_fir_fft_cc --fifo <fifo_path> <transition_bw> [window]`
+	bandpass_fir_fft_cc --fifo <fifo_path> <transition_bw> [window]
 
 By writing to the given FIFO file with the syntax below, you can control the shift rate:
 
@@ -327,7 +328,8 @@ E.g. you can send `-0.05 0.02\n`
 
 Licensing
 ---------
-Before the implementation of some algoritms, GPL-licensed code from other applications have been reviewed.
+Most of the code for `libcsdr` is under BSD license.  
+[link](#licensing) However, before the implementation of some algoritms, GPL-licensed code from other applications have been reviewed.
 In order to eliminate any licesing issues, these parts are placed under a different file.
 However, the library is still fully functional with BSD-only code, altough having only less-optimized versions of some algorithms.  
 It should also be noted that if you compile with `-DUSE_FFTW` and `-DLIBCSDR_GPL` (as default), the GPL license would apply on the whole result.