Reorder some global variables
This commit is contained in:
Chris Robinson
+188
-188
@@ -33,6 +33,16 @@
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#include "alError.h"
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#include "mixer_defs.h"
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/* This is a user config option for modifying the overall output of the reverb
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* effect.
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*/
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ALfloat ReverbBoost = 1.0f;
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/* Specifies whether to use a standard reverb effect in place of EAX reverb (no
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* high-pass, modulation, or echo).
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*/
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ALboolean EmulateEAXReverb = AL_FALSE;
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/* This is the maximum number of samples processed for each inner loop
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* iteration. */
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#define MAX_UPDATE_SAMPLES 256
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@@ -58,6 +68,184 @@ static void init_mixfunc(void)
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MixRowSamples = SelectRowMixer();
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}
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/* The B-Format to A-Format conversion matrix. The arrangement of rows is
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* deliberately chosen to align the resulting lines to their spatial opposites
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* (0:above front left <-> 3:above back right, 1:below front right <-> 2:below
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* back left). It's not quite opposite, since the A-Format results in a
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* tetrahedron, but it's close enough. Should the model be extended to 8-lines
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* in the future, true opposites can be used.
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*/
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static const aluMatrixf B2A = {{
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{ 0.288675134595f, 0.288675134595f, 0.288675134595f, 0.288675134595f },
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{ 0.288675134595f, -0.288675134595f, -0.288675134595f, 0.288675134595f },
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{ 0.288675134595f, 0.288675134595f, -0.288675134595f, -0.288675134595f },
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{ 0.288675134595f, -0.288675134595f, 0.288675134595f, -0.288675134595f }
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}};
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/* Converts A-Format to B-Format. */
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static const aluMatrixf A2B = {{
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{ 0.866025403785f, 0.866025403785f, 0.866025403785f, 0.866025403785f },
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{ 0.866025403785f, -0.866025403785f, 0.866025403785f, -0.866025403785f },
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{ 0.866025403785f, -0.866025403785f, -0.866025403785f, 0.866025403785f },
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{ 0.866025403785f, 0.866025403785f, -0.866025403785f, -0.866025403785f }
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}};
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static const ALfloat FadeStep = 1.0f / FADE_SAMPLES;
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/* The all-pass and delay lines have a variable length dependent on the
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* effect's density parameter. The resulting density multiplier is:
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*
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* multiplier = 1 + (density * LINE_MULTIPLIER)
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*
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* Thus the line multiplier below will result in a maximum density multiplier
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* of 10.
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*/
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static const ALfloat LINE_MULTIPLIER = 9.0f;
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/* All delay line lengths are specified in seconds.
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*
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* To approximate early reflections, we break them up into primary (those
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* arriving from the same direction as the source) and secondary (those
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* arriving from the opposite direction).
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*
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* The early taps decorrelate the 4-channel signal to approximate an average
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* room response for the primary reflections after the initial early delay.
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*
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* Given an average room dimension (d_a) and the speed of sound (c) we can
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* calculate the average reflection delay (r_a) regardless of listener and
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* source positions as:
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*
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* r_a = d_a / c
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* c = 343.3
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*
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* This can extended to finding the average difference (r_d) between the
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* maximum (r_1) and minimum (r_0) reflection delays:
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*
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* r_0 = 2 / 3 r_a
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* = r_a - r_d / 2
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* = r_d
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* r_1 = 4 / 3 r_a
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* = r_a + r_d / 2
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* = 2 r_d
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* r_d = 2 / 3 r_a
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* = r_1 - r_0
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*
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* As can be determined by integrating the 1D model with a source (s) and
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* listener (l) positioned across the dimension of length (d_a):
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*
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* r_d = int_(l=0)^d_a (int_(s=0)^d_a |2 d_a - 2 (l + s)| ds) dl / c
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*
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* The initial taps (T_(i=0)^N) are then specified by taking a power series
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* that ranges between r_0 and half of r_1 less r_0:
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*
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* R_i = 2^(i / (2 N - 1)) r_d
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* = r_0 + (2^(i / (2 N - 1)) - 1) r_d
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* = r_0 + T_i
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* T_i = R_i - r_0
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* = (2^(i / (2 N - 1)) - 1) r_d
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*
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* Assuming an average of 5m (up to 50m with the density multiplier), we get
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* the following taps:
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*/
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static const ALfloat EARLY_TAP_LENGTHS[4] =
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{
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0.000000e+0f, 1.010676e-3f, 2.126553e-3f, 3.358580e-3f
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};
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/* The early all-pass filter lengths are based on the early tap lengths:
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*
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* A_i = R_i / a
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*
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* Where a is the approximate maximum all-pass cycle limit (20).
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*/
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static const ALfloat EARLY_ALLPASS_LENGTHS[4] =
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{
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4.854840e-4f, 5.360178e-4f, 5.918117e-4f, 6.534130e-4f
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};
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/* The early delay lines are used to transform the primary reflections into
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* the secondary reflections. The A-format is arranged in such a way that
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* the channels/lines are spatially opposite:
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*
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* C_i is opposite C_(N-i-1)
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*
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* The delays of the two opposing reflections (R_i and O_i) from a source
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* anywhere along a particular dimension always sum to twice its full delay:
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*
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* 2 r_a = R_i + O_i
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*
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* With that in mind we can determine the delay between the two reflections
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* and thus specify our early line lengths (L_(i=0)^N) using:
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*
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* O_i = 2 r_a - R_(N-i-1)
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* L_i = O_i - R_(N-i-1)
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* = 2 (r_a - R_(N-i-1))
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* = 2 (r_a - T_(N-i-1) - r_0)
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* = 2 r_a (1 - (2 / 3) 2^((N - i - 1) / (2 N - 1)))
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*
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* Using an average dimension of 5m, we get:
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*/
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static const ALfloat EARLY_LINE_LENGTHS[4] =
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{
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2.992520e-3f, 5.456575e-3f, 7.688329e-3f, 9.709681e-3f
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};
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/* The late all-pass filter lengths are based on the late line lengths:
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*
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* A_i = (5 / 3) L_i / r_1
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*/
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static const ALfloat LATE_ALLPASS_LENGTHS[4] =
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{
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8.091400e-4f, 1.019453e-3f, 1.407968e-3f, 1.618280e-3f
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};
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/* The late lines are used to approximate the decaying cycle of recursive
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* late reflections.
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*
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* Splitting the lines in half, we start with the shortest reflection paths
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* (L_(i=0)^(N/2)):
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*
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* L_i = 2^(i / (N - 1)) r_d
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*
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* Then for the opposite (longest) reflection paths (L_(i=N/2)^N):
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*
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* L_i = 2 r_a - L_(i-N/2)
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* = 2 r_a - 2^((i - N / 2) / (N - 1)) r_d
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*
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* For our 5m average room, we get:
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*/
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static const ALfloat LATE_LINE_LENGTHS[4] =
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{
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9.709681e-3f, 1.223343e-2f, 1.689561e-2f, 1.941936e-2f
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};
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/* This coefficient is used to define the sinus depth according to the
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* modulation depth property. This value must be below half the shortest late
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* line length (0.0097/2 = ~0.0048), otherwise with certain parameters (high
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* mod time, low density) the downswing can sample before the input.
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*/
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static const ALfloat MODULATION_DEPTH_COEFF = 1.0f / 4096.0f;
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/* A filter is used to avoid the terrible distortion caused by changing
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* modulation time and/or depth. To be consistent across different sample
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* rates, the coefficient must be raised to a constant divided by the sample
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* rate: coeff^(constant / rate).
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*/
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static const ALfloat MODULATION_FILTER_COEFF = 0.048f;
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static const ALfloat MODULATION_FILTER_CONST = 100000.0f;
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/* Prior to VS2013, MSVC lacks the round() family of functions. */
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#if defined(_MSC_VER) && _MSC_VER < 1800
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static inline long lroundf(float val)
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{
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if(val < 0.0)
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return fastf2i(ceilf(val-0.5f));
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return fastf2i(floorf(val+0.5f));
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}
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#endif
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typedef struct DelayLineI {
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/* The delay lines use interleaved samples, with the lengths being powers
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* of 2 to allow the use of bit-masking instead of a modulus for wrapping.
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@@ -298,194 +486,6 @@ static ALvoid ALreverbState_Destruct(ALreverbState *State)
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ALeffectState_Destruct(STATIC_CAST(ALeffectState,State));
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}
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/* The B-Format to A-Format conversion matrix. The arrangement of rows is
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* deliberately chosen to align the resulting lines to their spatial opposites
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* (0:above front left <-> 3:above back right, 1:below front right <-> 2:below
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* back left). It's not quite opposite, since the A-Format results in a
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* tetrahedron, but it's close enough. Should the model be extended to 8-lines
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* in the future, true opposites can be used.
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*/
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static const aluMatrixf B2A = {{
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{ 0.288675134595f, 0.288675134595f, 0.288675134595f, 0.288675134595f },
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{ 0.288675134595f, -0.288675134595f, -0.288675134595f, 0.288675134595f },
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{ 0.288675134595f, 0.288675134595f, -0.288675134595f, -0.288675134595f },
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{ 0.288675134595f, -0.288675134595f, 0.288675134595f, -0.288675134595f }
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}};
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/* Converts A-Format to B-Format. */
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static const aluMatrixf A2B = {{
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{ 0.866025403785f, 0.866025403785f, 0.866025403785f, 0.866025403785f },
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{ 0.866025403785f, -0.866025403785f, 0.866025403785f, -0.866025403785f },
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{ 0.866025403785f, -0.866025403785f, -0.866025403785f, 0.866025403785f },
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{ 0.866025403785f, 0.866025403785f, -0.866025403785f, -0.866025403785f }
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}};
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static const ALfloat FadeStep = 1.0f / FADE_SAMPLES;
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/* This is a user config option for modifying the overall output of the reverb
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* effect.
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*/
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ALfloat ReverbBoost = 1.0f;
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/* Specifies whether to use a standard reverb effect in place of EAX reverb (no
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* high-pass, modulation, or echo).
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*/
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ALboolean EmulateEAXReverb = AL_FALSE;
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/* The all-pass and delay lines have a variable length dependent on the
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* effect's density parameter. The resulting density multiplier is:
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*
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* multiplier = 1 + (density * LINE_MULTIPLIER)
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*
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* Thus the line multiplier below will result in a maximum density multiplier
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* of 10.
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*/
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static const ALfloat LINE_MULTIPLIER = 9.0f;
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/* All delay line lengths are specified in seconds.
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*
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* To approximate early reflections, we break them up into primary (those
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* arriving from the same direction as the source) and secondary (those
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* arriving from the opposite direction).
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*
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* The early taps decorrelate the 4-channel signal to approximate an average
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* room response for the primary reflections after the initial early delay.
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*
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* Given an average room dimension (d_a) and the speed of sound (c) we can
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* calculate the average reflection delay (r_a) regardless of listener and
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* source positions as:
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*
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* r_a = d_a / c
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* c = 343.3
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*
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* This can extended to finding the average difference (r_d) between the
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* maximum (r_1) and minimum (r_0) reflection delays:
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*
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* r_0 = 2 / 3 r_a
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* = r_a - r_d / 2
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* = r_d
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* r_1 = 4 / 3 r_a
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* = r_a + r_d / 2
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* = 2 r_d
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* r_d = 2 / 3 r_a
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* = r_1 - r_0
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*
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* As can be determined by integrating the 1D model with a source (s) and
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* listener (l) positioned across the dimension of length (d_a):
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*
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* r_d = int_(l=0)^d_a (int_(s=0)^d_a |2 d_a - 2 (l + s)| ds) dl / c
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*
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* The initial taps (T_(i=0)^N) are then specified by taking a power series
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* that ranges between r_0 and half of r_1 less r_0:
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*
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* R_i = 2^(i / (2 N - 1)) r_d
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* = r_0 + (2^(i / (2 N - 1)) - 1) r_d
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* = r_0 + T_i
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* T_i = R_i - r_0
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* = (2^(i / (2 N - 1)) - 1) r_d
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*
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* Assuming an average of 5m (up to 50m with the density multiplier), we get
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* the following taps:
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*/
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static const ALfloat EARLY_TAP_LENGTHS[4] =
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{
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0.000000e+0f, 1.010676e-3f, 2.126553e-3f, 3.358580e-3f
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};
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/* The early all-pass filter lengths are based on the early tap lengths:
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*
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* A_i = R_i / a
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*
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* Where a is the approximate maximum all-pass cycle limit (20).
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*/
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static const ALfloat EARLY_ALLPASS_LENGTHS[4] =
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{
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4.854840e-4f, 5.360178e-4f, 5.918117e-4f, 6.534130e-4f
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};
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/* The early delay lines are used to transform the primary reflections into
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* the secondary reflections. The A-format is arranged in such a way that
|
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* the channels/lines are spatially opposite:
|
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*
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* C_i is opposite C_(N-i-1)
|
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*
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* The delays of the two opposing reflections (R_i and O_i) from a source
|
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* anywhere along a particular dimension always sum to twice its full delay:
|
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*
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* 2 r_a = R_i + O_i
|
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*
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* With that in mind we can determine the delay between the two reflections
|
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* and thus specify our early line lengths (L_(i=0)^N) using:
|
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*
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* O_i = 2 r_a - R_(N-i-1)
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* L_i = O_i - R_(N-i-1)
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* = 2 (r_a - R_(N-i-1))
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* = 2 (r_a - T_(N-i-1) - r_0)
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* = 2 r_a (1 - (2 / 3) 2^((N - i - 1) / (2 N - 1)))
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*
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* Using an average dimension of 5m, we get:
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*/
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static const ALfloat EARLY_LINE_LENGTHS[4] =
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{
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2.992520e-3f, 5.456575e-3f, 7.688329e-3f, 9.709681e-3f
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};
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/* The late all-pass filter lengths are based on the late line lengths:
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*
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* A_i = (5 / 3) L_i / r_1
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*/
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static const ALfloat LATE_ALLPASS_LENGTHS[4] =
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{
|
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8.091400e-4f, 1.019453e-3f, 1.407968e-3f, 1.618280e-3f
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};
|
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|
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/* The late lines are used to approximate the decaying cycle of recursive
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* late reflections.
|
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*
|
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* Splitting the lines in half, we start with the shortest reflection paths
|
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* (L_(i=0)^(N/2)):
|
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*
|
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* L_i = 2^(i / (N - 1)) r_d
|
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*
|
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* Then for the opposite (longest) reflection paths (L_(i=N/2)^N):
|
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*
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* L_i = 2 r_a - L_(i-N/2)
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* = 2 r_a - 2^((i - N / 2) / (N - 1)) r_d
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*
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* For our 5m average room, we get:
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*/
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static const ALfloat LATE_LINE_LENGTHS[4] =
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{
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9.709681e-3f, 1.223343e-2f, 1.689561e-2f, 1.941936e-2f
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};
|
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|
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/* This coefficient is used to define the sinus depth according to the
|
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* modulation depth property. This value must be below half the shortest late
|
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* line length (0.0097/2 = ~0.0048), otherwise with certain parameters (high
|
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* mod time, low density) the downswing can sample before the input.
|
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*/
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static const ALfloat MODULATION_DEPTH_COEFF = 1.0f / 4096.0f;
|
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|
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/* A filter is used to avoid the terrible distortion caused by changing
|
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* modulation time and/or depth. To be consistent across different sample
|
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* rates, the coefficient must be raised to a constant divided by the sample
|
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* rate: coeff^(constant / rate).
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*/
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static const ALfloat MODULATION_FILTER_COEFF = 0.048f;
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static const ALfloat MODULATION_FILTER_CONST = 100000.0f;
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|
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/* Prior to VS2013, MSVC lacks the round() family of functions. */
|
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#if defined(_MSC_VER) && _MSC_VER < 1800
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static inline long lroundf(float val)
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{
|
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if(val < 0.0)
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return fastf2i(ceilf(val-0.5f));
|
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return fastf2i(floorf(val+0.5f));
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}
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#endif
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/**************************************
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* Device Update *
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**************************************/
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Reference in New Issue
Block a user