Compressor: clarify and expand comments

pull/5/head
Chip Audette 8 years ago
parent 21fe4b37dc
commit 163877b76c
  1. 37
      AudioEffectCompressor_F32.h

@ -37,7 +37,7 @@ class AudioEffectCompressor_F32 : public AudioStream_F32
//apply a high-pass filter to get rid of the DC offset
if (use_HP_prefilter) arm_biquad_cascade_df1_f32(&hp_filt_struct, audio_block->data, audio_block->data, audio_block->length);
//apply the pre-gain...a negative gain value will disable
if (pre_gain > 0.0f) arm_scale_f32(audio_block->data, pre_gain, audio_block->data, audio_block->length); //use ARM DSP for speed!
@ -52,29 +52,32 @@ class AudioEffectCompressor_F32 : public AudioStream_F32
//apply the desired gain...store the processed audio back into audio_block
arm_mult_f32(audio_block->data, gain_block->data, audio_block->data, audio_block->length);
///transmit the block and release memory
//transmit the block and release memory
AudioStream_F32::transmit(audio_block);
AudioStream_F32::release(audio_block);
AudioStream_F32::release(gain_block);
AudioStream_F32::release(audio_level_dB_block);
}
// Here's the method that estimates the level of the audio (in dB)
// It squares the signal and low-pass filters to get a time-averaged
// signal power. It then
void calcAudioLevel_dB(audio_block_f32_t *wav_block, audio_block_f32_t *level_dB_block) {
//calculate the instantaneous signal power (square the signal)
// calculate the instantaneous signal power (square the signal)
audio_block_f32_t *wav_pow_block = AudioStream_F32::allocate_f32();
arm_mult_f32(wav_block->data, wav_block->data, wav_pow_block->data, wav_block->length);
//apply a smoothing filter to wav_pow_block and convert to dB
float c1 = level_lp_const;
float c2 = 1.0f - c1;
// low-pass filter and convert to dB
float c1 = level_lp_const, c2 = 1.0f - c1; //prepare constants
for (int i = 0; i < wav_pow_block->length; i++) {
//compute target gain (well, we're actualy calculating gain^2) assuming we want to copress
// first-order low-pass filter to get a running estimate of the average power
wav_pow_block->data[i] = c1*prev_level_lp_pow + c2*wav_pow_block->data[i];
// save the state of the first-order low-pass filter
prev_level_lp_pow = wav_pow_block->data[i];
//save state for next time (and for next data block)
prev_level_lp_pow = wav_pow_block->data[i];
//convert to dB (but not yet multiplied by 10.0
//now convert the signal power to dB (but not yet multiplied by 10.0)
level_dB_block->data[i] = log10f(wav_pow_block->data[i]);
}
@ -89,9 +92,11 @@ class AudioEffectCompressor_F32 : public AudioStream_F32
return; //output is passed through level_dB_block
}
//This method computes the desired gain from the compressor, given an estimate
//of the signal level (in dB)
void calcGain(audio_block_f32_t *audio_level_dB_block, audio_block_f32_t *gain_block) {
//first, calculate the instantaneous target gain
//first, calculate the instantaneous target gain based on the compression ratio
audio_block_f32_t *inst_targ_gain_dB_block = AudioStream_F32::allocate_f32();
calcInstantaneousTargetGain(audio_level_dB_block, inst_targ_gain_dB_block);
@ -99,9 +104,9 @@ class AudioEffectCompressor_F32 : public AudioStream_F32
audio_block_f32_t *gain_dB_block = AudioStream_F32::allocate_f32();
calcSmoothedGain_dB(inst_targ_gain_dB_block,gain_dB_block);
//finally, convert from dB to linear gain: gain = 10^(gain_dB/20)
//finally, convert from dB to linear gain: gain = 10^(gain_dB/20); (ie this takes care of the sqrt, too!)
arm_scale_f32(gain_dB_block->data, 1.0f/20.0f, gain_dB_block->data, gain_dB_block->length); //divide by 20
for (int i = 0; i < gain_dB_block->length; i++) gain_block->data[i] = powf(10.0f,gain_dB_block->data[i]); //10^(x)
for (int i = 0; i < gain_dB_block->length; i++) gain_block->data[i] = powf(10.0f,gain_dB_block->data[i]); //do the 10^(x)
//release memory and return
AudioStream_F32::release(gain_dB_block);
@ -109,6 +114,8 @@ class AudioEffectCompressor_F32 : public AudioStream_F32
return; //output is passed through gain_block
}
//Compute the instantaneous desired gain, including the compression ratio and
//threshold for where the comrpession kicks in
void calcInstantaneousTargetGain(audio_block_f32_t *audio_level_dB_block, audio_block_f32_t *inst_targ_gain_dB_block) {
// how much are we above the compression threshold?
@ -140,6 +147,8 @@ class AudioEffectCompressor_F32 : public AudioStream_F32
return; //output is passed through inst_targ_gain_dB_block
}
//this method applies the "attack" and "release" constants to smooth the
//target gain level through time.
void calcSmoothedGain_dB(audio_block_f32_t *inst_targ_gain_dB_block, audio_block_f32_t *gain_dB_block) {
float32_t gain_dB;
float32_t one_minus_attack_const = 1.0f - attack_const;

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