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author | RĂ©mi Verschelde <remi@verschelde.fr> | 2021-02-01 20:55:25 +0100 |
---|---|---|
committer | GitHub <noreply@github.com> | 2021-02-01 20:55:25 +0100 |
commit | d2e121650476af0a91bc24ea38253428e97f9695 (patch) | |
tree | 22b120a76f0430faa9e4a9383f7c0c83557616b6 /servers/audio | |
parent | 264504d76d2c1ea374d5508e6a2634e2b5002d84 (diff) | |
parent | 1d5042c9e265219dec8da7311879f12ef3ef698b (diff) |
Merge pull request #37547 from aaronfranke/tau
Use Math_TAU and deg2rad/etc in more places and optimize code
Diffstat (limited to 'servers/audio')
-rw-r--r-- | servers/audio/audio_filter_sw.cpp | 6 | ||||
-rw-r--r-- | servers/audio/effects/audio_effect_chorus.cpp | 4 | ||||
-rw-r--r-- | servers/audio/effects/audio_effect_delay.cpp | 2 | ||||
-rw-r--r-- | servers/audio/effects/audio_effect_distortion.cpp | 4 | ||||
-rw-r--r-- | servers/audio/effects/audio_effect_phaser.cpp | 6 | ||||
-rw-r--r-- | servers/audio/effects/audio_effect_spectrum_analyzer.cpp | 3 | ||||
-rw-r--r-- | servers/audio/effects/eq.cpp | 4 | ||||
-rw-r--r-- | servers/audio/effects/reverb.cpp | 4 |
8 files changed, 17 insertions, 16 deletions
diff --git a/servers/audio/audio_filter_sw.cpp b/servers/audio/audio_filter_sw.cpp index 580e061496..bcfa4c4c37 100644 --- a/servers/audio/audio_filter_sw.cpp +++ b/servers/audio/audio_filter_sw.cpp @@ -58,7 +58,7 @@ void AudioFilterSW::prepare_coefficients(Coeffs *p_coeffs) { final_cutoff = 1; //don't allow less than this } - double omega = 2.0 * Math_PI * final_cutoff / sampling_rate; + double omega = Math_TAU * final_cutoff / sampling_rate; double sin_v = Math::sin(omega); double cos_v = Math::cos(omega); @@ -132,7 +132,7 @@ void AudioFilterSW::prepare_coefficients(Coeffs *p_coeffs) { double hicutoff = resonance; double centercutoff = (cutoff + resonance) / 2.0; double bandwidth = (Math::log(centercutoff) - Math::log(hicutoff)) / Math::log((double)2); - omega = 2.0 * Math_PI * centercutoff / sampling_rate; + omega = Math_TAU * centercutoff / sampling_rate; alpha = Math::sin(omega) * Math::sinh(Math::log((double)2) / 2 * bandwidth * omega / Math::sin(omega)); a0 = 1 + alpha; @@ -197,7 +197,7 @@ void AudioFilterSW::set_stages(int p_stages) { //adjust for multiple stages /* Fouriertransform kernel to obtain response */ float AudioFilterSW::get_response(float p_freq, Coeffs *p_coeffs) { - float freq = p_freq / sampling_rate * Math_PI * 2.0f; + float freq = p_freq / sampling_rate * Math_TAU; float cx = p_coeffs->b0, cy = 0.0; diff --git a/servers/audio/effects/audio_effect_chorus.cpp b/servers/audio/effects/audio_effect_chorus.cpp index 1542273a24..76a995eb37 100644 --- a/servers/audio/effects/audio_effect_chorus.cpp +++ b/servers/audio/effects/audio_effect_chorus.cpp @@ -84,7 +84,7 @@ void AudioEffectChorusInstance::_process_chunk(const AudioFrame *p_src_frames, A if (v.cutoff == 0) { continue; } - float auxlp = expf(-2.0 * Math_PI * v.cutoff / mix_rate); + float auxlp = expf(-Math_TAU * v.cutoff / mix_rate); float c1 = 1.0 - auxlp; float c2 = auxlp; AudioFrame h = filter_h[vc]; @@ -104,7 +104,7 @@ void AudioEffectChorusInstance::_process_chunk(const AudioFrame *p_src_frames, A float phase = (float)(local_cycles & AudioEffectChorus::CYCLES_MASK) / (float)(1 << AudioEffectChorus::CYCLES_FRAC); - float wave_delay = sinf(phase * 2.0 * Math_PI) * max_depth_frames; + float wave_delay = sinf(phase * Math_TAU) * max_depth_frames; int wave_delay_frames = lrint(floor(wave_delay)); float wave_delay_frac = wave_delay - (float)wave_delay_frames; diff --git a/servers/audio/effects/audio_effect_delay.cpp b/servers/audio/effects/audio_effect_delay.cpp index f04ab45ec1..ba50eeb0a3 100644 --- a/servers/audio/effects/audio_effect_delay.cpp +++ b/servers/audio/effects/audio_effect_delay.cpp @@ -75,7 +75,7 @@ void AudioEffectDelayInstance::_process_chunk(const AudioFrame *p_src_frames, Au tap2_vol.r *= CLAMP(1.0 + base->tap_2_pan, 0, 1); // feedback lowpass here - float lpf_c = expf(-2.0 * Math_PI * base->feedback_lowpass / mix_rate); // 0 .. 10khz + float lpf_c = expf(-Math_TAU * base->feedback_lowpass / mix_rate); // 0 .. 10khz float lpf_ic = 1.0 - lpf_c; const AudioFrame *src = p_src_frames; diff --git a/servers/audio/effects/audio_effect_distortion.cpp b/servers/audio/effects/audio_effect_distortion.cpp index 8f713ace22..b79434e7c2 100644 --- a/servers/audio/effects/audio_effect_distortion.cpp +++ b/servers/audio/effects/audio_effect_distortion.cpp @@ -36,8 +36,8 @@ void AudioEffectDistortionInstance::process(const AudioFrame *p_src_frames, Audi const float *src = (const float *)p_src_frames; float *dst = (float *)p_dst_frames; - //float lpf_c=expf(-2.0*Math_PI*keep_hf_hz.get()/(mix_rate*(float)OVERSAMPLE)); - float lpf_c = expf(-2.0 * Math_PI * base->keep_hf_hz / (AudioServer::get_singleton()->get_mix_rate())); + //float lpf_c=expf(-Math_TAU*keep_hf_hz.get()/(mix_rate*(float)OVERSAMPLE)); + float lpf_c = expf(-Math_TAU * base->keep_hf_hz / (AudioServer::get_singleton()->get_mix_rate())); float lpf_ic = 1.0 - lpf_c; float drive_f = base->drive; diff --git a/servers/audio/effects/audio_effect_phaser.cpp b/servers/audio/effects/audio_effect_phaser.cpp index 5e4e183ccf..9b70f03a19 100644 --- a/servers/audio/effects/audio_effect_phaser.cpp +++ b/servers/audio/effects/audio_effect_phaser.cpp @@ -38,13 +38,13 @@ void AudioEffectPhaserInstance::process(const AudioFrame *p_src_frames, AudioFra float dmin = base->range_min / (sampling_rate / 2.0); float dmax = base->range_max / (sampling_rate / 2.0); - float increment = 2.f * Math_PI * (base->rate / sampling_rate); + float increment = Math_TAU * (base->rate / sampling_rate); for (int i = 0; i < p_frame_count; i++) { phase += increment; - while (phase >= Math_PI * 2.f) { - phase -= Math_PI * 2.f; + while (phase >= Math_TAU) { + phase -= Math_TAU; } float d = dmin + (dmax - dmin) * ((sin(phase) + 1.f) / 2.f); diff --git a/servers/audio/effects/audio_effect_spectrum_analyzer.cpp b/servers/audio/effects/audio_effect_spectrum_analyzer.cpp index 7f73f2e880..3f7ab74a74 100644 --- a/servers/audio/effects/audio_effect_spectrum_analyzer.cpp +++ b/servers/audio/effects/audio_effect_spectrum_analyzer.cpp @@ -110,10 +110,11 @@ void AudioEffectSpectrumAnalyzerInstance::process(const AudioFrame *p_src_frames while (p_frame_count) { int to_fill = fft_size * 2 - temporal_fft_pos; to_fill = MIN(to_fill, p_frame_count); + const double to_fill_step = Math_TAU / (double)fft_size; float *fftw = temporal_fft.ptrw(); for (int i = 0; i < to_fill; i++) { //left and right buffers - float window = -0.5 * Math::cos(2.0 * Math_PI * (double)temporal_fft_pos / (double)fft_size) + 0.5; + float window = -0.5 * Math::cos(to_fill_step * (double)temporal_fft_pos) + 0.5; fftw[temporal_fft_pos * 2] = window * p_src_frames->l; fftw[temporal_fft_pos * 2 + 1] = 0; fftw[(temporal_fft_pos + fft_size * 2) * 2] = window * p_src_frames->r; diff --git a/servers/audio/effects/eq.cpp b/servers/audio/effects/eq.cpp index 2181411b9e..e0c3eb6d3a 100644 --- a/servers/audio/effects/eq.cpp +++ b/servers/audio/effects/eq.cpp @@ -89,8 +89,8 @@ void EQ::recalculate_band_coefficients() { double frq_l = round(frq / pow(2.0, octave_size / 2.0)); double side_gain2 = POW2(Math_SQRT12); - double th = 2.0 * Math_PI * frq / mix_rate; - double th_l = 2.0 * Math_PI * frq_l / mix_rate; + double th = Math_TAU * frq / mix_rate; + double th_l = Math_TAU * frq_l / mix_rate; double c2a = side_gain2 * POW2(cos(th)) - 2.0 * side_gain2 * cos(th_l) * cos(th) + side_gain2 - POW2(sin(th_l)); diff --git a/servers/audio/effects/reverb.cpp b/servers/audio/effects/reverb.cpp index eb96e21659..7df2f99f67 100644 --- a/servers/audio/effects/reverb.cpp +++ b/servers/audio/effects/reverb.cpp @@ -91,7 +91,7 @@ void Reverb::process(float *p_src, float *p_dst, int p_frames) { } if (params.hpf > 0) { - float hpaux = expf(-2.0 * Math_PI * params.hpf * 6000 / params.mix_rate); + float hpaux = expf(-Math_TAU * params.hpf * 6000 / params.mix_rate); float hp_a1 = (1.0 + hpaux) / 2.0; float hp_a2 = -(1.0 + hpaux) / 2.0; float hp_b1 = hpaux; @@ -293,7 +293,7 @@ void Reverb::update_parameters() { float auxdmp = params.damp / 2.0 + 0.5; //only half the range (0.5 .. 1.0 is enough) auxdmp *= auxdmp; - c.damp = expf(-2.0 * Math_PI * auxdmp * 10000 / params.mix_rate); // 0 .. 10khz + c.damp = expf(-Math_TAU * auxdmp * 10000 / params.mix_rate); // 0 .. 10khz } } |