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@@ -85,8 +85,8 @@ enum
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// ncurses is notoriously retarded for input handling, we need something
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// different if only to receive mouse events reliably.
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//
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// 2020 update: ncurses is mostly reliable now but rxvt-unicode needs to start
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// supporting 1006, or ncurses needs to start supporting the 1015 mode.
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// 2021 update: ncurses is mostly reliable now, though rxvt-unicode only
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// supports the 1006 mode that ncurses also supports mode starting with 9.25.
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#include "termo.h"
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@@ -95,6 +95,13 @@ enum
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#include <curl/curl.h>
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// The spectrum analyser requires a DFT transform. The FFTW library is fairly
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// efficient, and doesn't have a requirement on the number of bins.
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#ifdef WITH_FFTW
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#include <fftw3.h>
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#endif // WITH_FFTW
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#define APP_TITLE PROGRAM_NAME ///< Left top corner
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// --- Utilities ---------------------------------------------------------------
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@@ -560,6 +567,273 @@ item_list_resize (struct item_list *self, size_t len)
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self->len = len;
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}
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// --- Spectrum analyzer -------------------------------------------------------
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#ifdef WITH_FFTW
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struct spectrum
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{
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int sampling_rate; ///< Number of samples per seconds
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int channels; ///< Number of sampled channels
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int bits; ///< Number of bits per sample
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int bars; ///< Number of output vertical bars
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int bins; ///< Number of DFT bins
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int useful_bins; ///< Bins up to the Nyquist frequency
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int samples; ///< Number of windows to average
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float accumulator_scale; ///< Scaling factor for accum. values
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int *top_bins; ///< Top DFT bin index for each bar
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char *spectrum; ///< String buffer for the "render"
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void *buffer; ///< Input buffer
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size_t buffer_len; ///< Input buffer fill level
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size_t buffer_size; ///< Input buffer size
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/// Decode the respective part of the buffer into the last 1/3 of data
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void (*decode) (struct spectrum *, int sample);
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float *data; ///< Normalized audio data
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float *window; ///< Sampled window function
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float *windowed; ///< data * window
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fftwf_complex *out; ///< DFT output
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fftwf_plan p; ///< DFT plan/FFTW configuration
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float *accumulator; ///< Accumulated powers of samples
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};
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// - - Windows - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
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// Out: float[n] of 0..1
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static void
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window_hann (float *coefficients, size_t n)
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{
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for (size_t i = 0; i < n; i++)
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{
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float sine = sin (M_PI * i / n);
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coefficients[i] = sine * sine;
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}
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}
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// In: float[n] of -1..1, float[n] of 0..1; out: float[n] of -1..1
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static void
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window_apply (const float *in, const float *coefficients, float *out, size_t n)
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{
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for (size_t i = 0; i < n; i++)
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out[i] = in[i] * coefficients[i];
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}
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// In: float[n] of 0..1; out: float 0..n, describing the coherent gain
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static float
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window_coherent_gain (const float *in, size_t n)
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{
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float sum = 0;
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for (size_t i = 0; i < n; i++)
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sum += in[i];
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return sum;
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}
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// - - Decoding - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
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static void
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spectrum_decode_8 (struct spectrum *s, int sample)
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{
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size_t n = s->useful_bins;
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float *data = s->data + n;
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int8_t *p = (int8_t *) s->buffer + sample * n * s->channels;
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while (n--)
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{
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int32_t acc = 0;
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for (int ch = 0; ch < s->channels; ch++)
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acc += *p++;
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*data++ = (float) acc / -INT8_MIN / s->channels;
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}
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}
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static void
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spectrum_decode_16 (struct spectrum *s, int sample)
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{
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size_t n = s->useful_bins;
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float *data = s->data + n;
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int16_t *p = (int16_t *) s->buffer + sample * n * s->channels;
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while (n--)
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{
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int32_t acc = 0;
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for (int ch = 0; ch < s->channels; ch++)
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acc += *p++;
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*data++ = (float) acc / -INT16_MIN / s->channels;
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}
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}
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// - - Spectrum analysis - - - - - - - - - - - - - - - - - - - - - - - - - - - -
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static const char *spectrum_bars[] =
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{ " ", "▁", "▂", "▃", "▄", "▅", "▆", "▇", "█" };
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/// Assuming the input buffer is full, updates the rendered spectrum
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static void
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spectrum_sample (struct spectrum *s)
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{
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memset (s->accumulator, 0, sizeof *s->accumulator * s->useful_bins);
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// Credit for the algorithm goes to Audacity's /src/SpectrumAnalyst.cpp,
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// apparently Welch's method
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for (int sample = 0; sample < s->samples; sample++)
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{
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// We use 50% overlap and start with data from the last run (if any)
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memmove (s->data, s->data + s->useful_bins,
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sizeof *s->data * s->useful_bins);
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s->decode (s, sample);
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window_apply (s->data, s->window, s->windowed, s->bins);
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fftwf_execute (s->p);
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for (int bin = 0; bin < s->useful_bins; bin++)
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{
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// out[0][0] is the DC component, not useful to us
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float re = s->out[bin + 1][0];
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float im = s->out[bin + 1][1];
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s->accumulator[bin] += re * re + im * im;
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}
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}
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int last_bin = 0;
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char *p = s->spectrum;
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for (int bar = 0; bar < s->bars; bar++)
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{
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int top_bin = s->top_bins[bar];
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// Think of this as accumulating energies within bands,
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// so that it matches our non-linear hearing--there's no averaging.
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// For more precision, we could employ an "equal loudness contour".
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float acc = 0;
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for (int bin = last_bin; bin < top_bin; bin++)
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acc += s->accumulator[bin];
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last_bin = top_bin;
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float db = 10 * log10f (acc * s->accumulator_scale);
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if (db > 0)
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db = 0;
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// Assuming decibels are always negative (i.e., properly normalized).
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// The division defines the cutoff: 9 * 7 = 63 dB of range.
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int height = N_ELEMENTS (spectrum_bars) - 1 + (int) (db / 7);
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p += strlen (strcpy (p, spectrum_bars[MAX (height, 0)]));
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}
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}
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static bool
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spectrum_init (struct spectrum *s, char *format, int bars, struct error **e)
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{
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errno = 0;
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long sampling_rate, bits, channels;
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if (!format
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|| (sampling_rate = strtol (format, &format, 10), *format++ != ':')
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|| (bits = strtol (format, &format, 10), *format++ != ':')
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|| (channels = strtol (format, &format, 10), *format)
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|| errno != 0)
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return error_set (e, "invalid format, expected RATE:BITS:CHANNELS");
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if (sampling_rate < 20000 || sampling_rate > INT_MAX)
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return error_set (e, "unsupported sampling rate (%ld)", sampling_rate);
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if (bits != 8 && bits != 16)
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return error_set (e, "unsupported bit count (%ld)", bits);
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if (channels < 1 || channels > INT_MAX)
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return error_set (e, "no channels to sample (%ld)", channels);
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if (bars < 1 || bars > 12)
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return error_set (e, "requested too few or too many bars (%d)", bars);
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// All that can fail henceforth is memory allocation
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*s = (struct spectrum)
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{
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.sampling_rate = sampling_rate,
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.bits = bits,
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.channels = channels,
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.bars = bars,
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};
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// The number of bars is always smaller than that of the samples (bins).
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// Let's start with the equation of the top FFT bin to use for a given bar:
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// top_bin = (num_bins + 1) ^ (bar / num_bars) - 1
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// N.b. if we didn't subtract, the power function would make this ≥ 1.
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// N.b. we then also need to extend the range by the same amount.
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//
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// We need the amount of bins for the first bar to be at least one:
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// 1 ≤ (num_bins + 1) ^ (1 / num_bars) - 1
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//
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// Solving with Wolfram Alpha gives us:
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// num_bins ≥ (2 ^ num_bars) - 1 [for y > 0]
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//
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// And we need to remember that half of the FFT bins are useless/missing--
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// FFTW skips useless points past the Nyquist frequency.
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int necessary_bins = 2 << s->bars;
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// Discard frequencies above 20 kHz, which take up a constant ratio
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// of all bins, given by the sampling rate. A more practical/efficient
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// solution would be to just handle 96/192/... kHz as bitshifts.
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//
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// Trying to filter out sub-20 Hz frequencies would be even more wasteful.
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double audible_ratio = s->sampling_rate / 2. / 20000;
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s->bins = ceil (necessary_bins * MAX (audible_ratio, 1));
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s->useful_bins = s->bins / 2;
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int used_bins = necessary_bins / 2;
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s->spectrum = xcalloc (sizeof *s->spectrum, s->bars * 3 + 1);
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s->top_bins = xcalloc (sizeof *s->top_bins, s->bars);
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for (int bar = 0; bar < s->bars; bar++)
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{
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int top_bin = floor (pow (used_bins + 1, (bar + 1.) / s->bars)) - 1;
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s->top_bins[bar] = MIN (top_bin, used_bins);
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}
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// Limit updates to 30 times per second to limit CPU load
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s->samples = s->sampling_rate / s->bins * 2 / 30;
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if (s->samples < 1)
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s->samples = 1;
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if (s->bits == 8) s->decode = spectrum_decode_8;
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if (s->bits == 16) s->decode = spectrum_decode_16;
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s->buffer_size = s->samples * s->useful_bins * s->bits / 8 * s->channels;
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s->buffer = xcalloc (1, s->buffer_size);
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// Prepare the window
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s->window = xcalloc (sizeof *s->window, s->bins);
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window_hann (s->window, s->bins);
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// Multiply by 2 for only using half of the DFT's result, then adjust to
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// the total energy of the window. Both squared, because the accumulator
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// contains squared values. Compute the average, and convert to decibels.
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// See also the mildly confusing https://dsp.stackexchange.com/a/14945.
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float coherent_gain = window_coherent_gain (s->window, s->bins);
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s->accumulator_scale = 2 * 2 / coherent_gain / coherent_gain / s->samples;
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s->data = xcalloc (sizeof *s->data, s->bins);
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s->windowed = fftw_malloc (sizeof *s->windowed * s->bins);
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s->out = fftw_malloc (sizeof *s->out * (s->useful_bins + 1));
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s->p = fftwf_plan_dft_r2c_1d (s->bins, s->windowed, s->out, FFTW_MEASURE);
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s->accumulator = xcalloc (sizeof *s->accumulator, s->useful_bins);
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return true;
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}
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static void
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spectrum_free (struct spectrum *s)
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{
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free (s->accumulator);
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fftwf_destroy_plan (s->p);
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fftw_free (s->out);
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fftw_free (s->windowed);
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free (s->data);
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free (s->window);
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free (s->spectrum);
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free (s->top_bins);
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free (s->buffer);
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memset (s, 0, sizeof *s);
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}
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#endif // WITH_FFTW
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// --- Application -------------------------------------------------------------
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// Function names are prefixed mostly because of curses which clutters the
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@@ -675,6 +949,13 @@ static struct app_context
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int gauge_offset; ///< Offset to the gauge or -1
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int gauge_width; ///< Width of the gauge, if present
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#ifdef WITH_FFTW
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struct spectrum spectrum; ///< Spectrum analyser
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int spectrum_fd; ///< FIFO file descriptor (non-blocking)
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int spectrum_column, spectrum_row; ///< Position for fast refresh
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struct poller_fd spectrum_event; ///< FIFO watcher
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#endif // WITH_FFTW
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struct line_editor editor; ///< Line editor
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struct poller_idle refresh_event; ///< Refresh the screen
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@@ -750,6 +1031,22 @@ static struct config_schema g_config_settings[] =
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.comment = "Where all the files MPD is playing are located",
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.type = CONFIG_ITEM_STRING },
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#ifdef WITH_FFTW
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{ .name = "spectrum_path",
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.comment = "Visualizer feed path to a FIFO audio output",
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.type = CONFIG_ITEM_STRING },
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// MPD's "outputs" command doesn't include this information
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{ .name = "spectrum_format",
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.comment = "Visualizer feed data format",
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.type = CONFIG_ITEM_STRING,
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.default_ = "\"44100:16:2\"" },
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// 10 is about the useful limit, then it gets too computationally expensive
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{ .name = "spectrum_bars",
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.comment = "Number of computed audio spectrum bars",
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.type = CONFIG_ITEM_INTEGER,
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.default_ = "8" },
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#endif // WITH_FFTW
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// Disabling this minimises MPD traffic and has the following caveats:
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// - when MPD stalls on retrieving audio data, we keep ticking
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// - when the "play" succeeds in ACTION_MPD_REPLACE for the same item as
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@@ -904,6 +1201,11 @@ app_init_context (void)
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g.playback_info = str_map_make (free);
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g.playback_info.key_xfrm = tolower_ascii_strxfrm;
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#ifdef WITH_FFTW
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g.spectrum_fd = -1;
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g.spectrum_row = g.spectrum_column = -1;
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#endif // WITH_FFTW
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// This is also approximately what libunistring does internally,
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// since the locale name is canonicalized by locale_charset().
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// Note that non-Unicode locales are handled pretty inefficiently.
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@@ -957,6 +1259,15 @@ app_free_context (void)
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strv_free (&g.streams);
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item_list_free (&g.playlist);
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#ifdef WITH_FFTW
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spectrum_free (&g.spectrum);
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if (g.spectrum_fd != -1)
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{
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poller_fd_reset (&g.spectrum_event);
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xclose (g.spectrum_fd);
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}
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#endif // WITH_FFTW
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line_editor_free (&g.editor);
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config_free (&g.config);
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@@ -1218,6 +1529,21 @@ app_draw_header (void)
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g.tabs_offset = g.header_height;
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LIST_FOR_EACH (struct tab, iter, g.tabs)
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row_buffer_append (&buf, iter->name, attrs[iter == g.active_tab]);
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#ifdef WITH_FFTW
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// This seems like the most reasonable, otherwise unoccupied space
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if (g.spectrum_fd != -1)
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{
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// Find some space and remember where it was, for fast refreshes
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row_buffer_ellipsis (&buf, COLS - g.spectrum.bars - 1);
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row_buffer_align (&buf, COLS - g.spectrum.bars, attrs[false]);
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g.spectrum_row = g.header_height;
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g.spectrum_column = buf.total_width;
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row_buffer_append (&buf, g.spectrum.spectrum, attrs[false]);
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}
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#endif // WITH_FFTW
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app_flush_header (&buf, attrs[false]);
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const char *header = g.active_tab->header;
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@@ -3421,6 +3747,137 @@ debug_tab_init (void)
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return super;
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}
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// --- Spectrum analyser -------------------------------------------------------
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#ifdef WITH_FFTW
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static void
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spectrum_redraw (void)
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{
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// A full refresh would be too computationally expensive,
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// let's hack around it in this case
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if (g.spectrum_row != -1)
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{
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attrset (APP_ATTR (TAB_BAR));
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mvaddstr (g.spectrum_row, g.spectrum_column, g.spectrum.spectrum);
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attrset (0);
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refresh ();
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}
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else
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app_invalidate ();
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}
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// When any problem occurs with the FIFO, we'll just give up on it completely
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static void
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spectrum_discard_fifo (void)
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{
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if (g.spectrum_fd != -1)
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{
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poller_fd_reset (&g.spectrum_event);
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xclose (g.spectrum_fd);
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g.spectrum_fd = -1;
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spectrum_free (&g.spectrum);
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g.spectrum_row = g.spectrum_column = -1;
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app_invalidate ();
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}
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}
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static void
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spectrum_on_fifo_readable (const struct pollfd *pfd, void *user_data)
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{
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(void) user_data;
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struct spectrum *s = &g.spectrum;
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bool update = false;
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ssize_t n;
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restart:
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while ((n = read (pfd->fd,
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s->buffer + s->buffer_len, s->buffer_size - s->buffer_len)) > 0)
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if ((s->buffer_len += n) == s->buffer_size)
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{
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update = true;
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spectrum_sample (s);
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s->buffer_len = 0;
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}
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if (!n)
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spectrum_discard_fifo ();
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else if (errno == EINTR)
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goto restart;
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else if (errno != EAGAIN)
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{
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print_error ("spectrum: %s", strerror (errno));
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spectrum_discard_fifo ();
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}
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else if (update)
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spectrum_redraw ();
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}
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// When playback is stopped, we need to feed the analyser some zeroes ourselves.
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// We could also just hide it. Hard to say which is simpler or better.
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static void
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spectrum_clear (void)
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{
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if (g.spectrum_fd != -1)
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{
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struct spectrum *s = &g.spectrum;
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memset (s->buffer, 0, s->buffer_size);
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spectrum_sample (s);
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spectrum_sample (s);
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s->buffer_len = 0;
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spectrum_redraw ();
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}
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}
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static void
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spectrum_setup_fifo (void)
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{
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const char *spectrum_path =
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get_config_string (g.config.root, "settings.spectrum_path");
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const char *spectrum_format =
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get_config_string (g.config.root, "settings.spectrum_format");
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struct config_item *spectrum_bars =
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config_item_get (g.config.root, "settings.spectrum_bars", NULL);
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if (!spectrum_path)
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return;
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struct error *e = NULL;
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char *path = resolve_filename
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(spectrum_path, resolve_relative_config_filename);
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if (!path)
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print_error ("spectrum: %s", "FIFO path could not be resolved");
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else if (!g.locale_is_utf8)
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print_error ("spectrum: %s", "UTF-8 locale required");
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else if (!spectrum_init (&g.spectrum,
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(char *) spectrum_format, spectrum_bars->value.integer, &e))
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{
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print_error ("spectrum: %s", e->message);
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error_free (e);
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}
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else if ((g.spectrum_fd = open (path, O_RDONLY | O_NONBLOCK)) == -1)
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{
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print_error ("spectrum: %s: %s", path, strerror (errno));
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spectrum_free (&g.spectrum);
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}
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else
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{
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g.spectrum_event = poller_fd_make (&g.poller, g.spectrum_fd);
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g.spectrum_event.dispatcher = spectrum_on_fifo_readable;
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poller_fd_set (&g.spectrum_event, POLLIN);
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}
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free (path);
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}
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#else // ! WITH_FFTW
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#define spectrum_setup_fifo()
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#define spectrum_clear()
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#define spectrum_discard_fifo()
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#endif // ! WITH_FFTW
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// --- MPD interface -----------------------------------------------------------
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static void
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@@ -3482,6 +3939,9 @@ mpd_update_playback_state (void)
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if (!strcmp (state, "pause")) g.state = PLAYER_PAUSED;
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}
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if (g.state == PLAYER_STOPPED)
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spectrum_clear ();
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// Values in "time" are always rounded. "elapsed", introduced in MPD 0.16,
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// is in millisecond precision and "duration" as well, starting with 0.20.
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// Prefer the more precise values but use what we have.
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@@ -3736,6 +4196,8 @@ mpd_on_connected (void *user_data)
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mpd_request_info ();
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library_tab_reload (NULL);
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}
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spectrum_setup_fifo ();
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}
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static void
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@@ -3752,6 +4214,8 @@ mpd_on_failure (void *user_data)
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mpd_update_playback_state ();
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current_tab_update ();
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info_tab_update ();
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spectrum_discard_fifo ();
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}
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static void
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