diff --git a/AudioConvert_F32.h b/AudioConvert_F32.h new file mode 100644 index 0000000..f0f3484 --- /dev/null +++ b/AudioConvert_F32.h @@ -0,0 +1,78 @@ + +#ifndef _AudioConvert_I16toF32 +#define _AudioConvert_I16toF32 + + +#include + +class AudioConvert_I16toF32 : public AudioStream_F32 //receive Int and transmits Float +{ + public: + AudioConvert_I16toF32(void) : AudioStream_F32(1, inputQueueArray_f32) { }; + void update(void) { + //get the Int16 block + audio_block_t *int_block; + int_block = AudioStream::receiveReadOnly(); //int16 data block + if (!int_block) return; + + //allocate a float block + audio_block_f32_t *float_block; + float_block = AudioStream_F32::allocate_f32(); + if (float_block == NULL) return; + + //convert to float + convertAudio_I16toF32(int_block, float_block, AUDIO_BLOCK_SAMPLES); + + //transmit the audio and return it to the system + AudioStream_F32::transmit(float_block,0); + AudioStream_F32::release(float_block); + AudioStream::release(int_block); + }; + + private: + audio_block_f32_t *inputQueueArray_f32[1]; //2 for stereo + + static void convertAudio_I16toF32(audio_block_t *in, audio_block_f32_t *out, int len) { + const float MAX_INT = 32678.0; + //for (int i = 0; i < len; i++) out->data[i] = (float)(in->data[i])/MAX_INT; + for (int i = 0; i < len; i++) out->data[i] = (float)(in->data[i]); + arm_scale_f32(out->data, 1.0/MAX_INT, out->data, out->length); //divide by 32678 to get -1.0 to +1.0 + } +}; + + +class AudioConvert_F32toI16 : public AudioStream_F32 //receive Float and transmits Int +{ + public: + AudioConvert_F32toI16(void) : AudioStream_F32(1, inputQueueArray_Float) {}; + void update(void) { + //get the float block + audio_block_f32_t *float_block; + float_block = AudioStream_F32::receiveReadOnly_f32(); //float data block + if (!float_block) return; + + //allocate a Int16 block + audio_block_t *int_block; + int_block = AudioStream::allocate(); + if (int_block == NULL) return; + + //convert back to int16 + convertAudio_F32ToI16(float_block, int_block, AUDIO_BLOCK_SAMPLES); + + //return audio to the system + AudioStream::transmit(int_block); + AudioStream::release(int_block); + AudioStream_F32::release(float_block); + }; + + private: + audio_block_f32_t *inputQueueArray_Float[1]; + static void convertAudio_F32ToI16(audio_block_f32_t *in, audio_block_t *out, int len) { + const float MAX_INT = 32678.0; + for (int i = 0; i < len; i++) { + out->data[i] = (int16_t)(max(min( (in->data[i] * MAX_INT), MAX_INT), -MAX_INT)); + } + } +}; + +#endif \ No newline at end of file diff --git a/AudioEffectGain_F32.cpp b/AudioEffectGain_F32.cpp new file mode 100644 index 0000000..e69de29 diff --git a/AudioEffectGain_F32.h b/AudioEffectGain_F32.h new file mode 100644 index 0000000..bc375a5 --- /dev/null +++ b/AudioEffectGain_F32.h @@ -0,0 +1,52 @@ +/* + * AudioEffectsGain + * + * Created: Chip Audette, November 2016 + * Purpose; Apply digital gain to the audio data. Assumes floating-point data. + * + * This processes a single stream fo audio data (ie, it is mono) + * + * MIT License. use at your own risk. +*/ + +#include //ARM DSP extensions. for speed! +#include + +class AudioEffectGain_F32 : public AudioStream_F32 //AudioStream_F32 is in AudioFloatProcessing.h +{ + public: + //constructor + AudioEffectGain_F32(void) : AudioStream_F32(1, inputQueueArray_f32) {}; + + //here's the method that does all the work + void update(void) { + //Serial.println("AudioEffectGain_F32: updating."); //for debugging. + audio_block_f32_t *block; + block = AudioStream_F32::receiveWritable_f32(); + if (!block) return; + + //apply the gain + //for (int i = 0; i < AUDIO_BLOCK_SAMPLES; i++) block->data[i] = gain * (block->data[i]); //non DSP way to do it + arm_scale_f32(block->data, gain, block->data, block->length); //use ARM DSP for speed! + + //transmit the block and be done + AudioStream_F32::transmit(block); + AudioStream_F32::release(block); + } + + //methods to set parameters of this module + void setGain(float g) { gain = g; } + void setGain_dB(float gain_dB) { + float gain = pow(10.0, gain_dB / 20.0); + setGain(gain); + } + + //methods to return information about this module + float getGain(void) { return gain; } + float getGain_dB(void) { return 20.0*log10(gain); } + + private: + audio_block_f32_t *inputQueueArray_f32[1]; //memory pointer for the input to this module + float gain = 1.0; //default value +}; + diff --git a/AudioStream_F32.cpp b/AudioStream_F32.cpp new file mode 100644 index 0000000..edf412d --- /dev/null +++ b/AudioStream_F32.cpp @@ -0,0 +1,156 @@ + +#include + +audio_block_f32_t * AudioStream_F32::f32_memory_pool; +uint32_t AudioStream_F32::f32_memory_pool_available_mask[6]; + +uint8_t AudioStream_F32::f32_memory_used = 0; +uint8_t AudioStream_F32::f32_memory_used_max = 0; + +// Set up the pool of audio data blocks +// placing them all onto the free list +void AudioStream_F32::initialize_f32_memory(audio_block_f32_t *data, unsigned int num) +{ + unsigned int i; + + //Serial.println("AudioStream_F32 initialize_memory"); + //delay(10); + if (num > 192) num = 192; + __disable_irq(); + f32_memory_pool = data; + for (i=0; i < 6; i++) { + f32_memory_pool_available_mask[i] = 0; + } + for (i=0; i < num; i++) { + f32_memory_pool_available_mask[i >> 5] |= (1 << (i & 0x1F)); + } + for (i=0; i < num; i++) { + data[i].memory_pool_index = i; + } + __enable_irq(); + +} // end initialize_memory + +// Allocate 1 audio data block. If successful +// the caller is the only owner of this new block +audio_block_f32_t * AudioStream_F32::allocate_f32(void) +{ + uint32_t n, index, avail; + uint32_t *p; + audio_block_f32_t *block; + uint8_t used; + + p = f32_memory_pool_available_mask; + __disable_irq(); + do { + avail = *p; if (avail) break; + p++; avail = *p; if (avail) break; + p++; avail = *p; if (avail) break; + p++; avail = *p; if (avail) break; + p++; avail = *p; if (avail) break; + p++; avail = *p; if (avail) break; + __enable_irq(); + //Serial.println("alloc_f32:null"); + return NULL; + } while (0); + n = __builtin_clz(avail); + *p = avail & ~(0x80000000 >> n); + used = f32_memory_used + 1; + f32_memory_used = used; + __enable_irq(); + index = p - f32_memory_pool_available_mask; + block = f32_memory_pool + ((index << 5) + (31 - n)); + block->ref_count = 1; + if (used > f32_memory_used_max) f32_memory_used_max = used; + //Serial.print("alloc_f32:"); + //Serial.println((uint32_t)block, HEX); + return block; +} + + +// Release ownership of a data block. If no +// other streams have ownership, the block is +// returned to the free pool +void AudioStream_F32::release(audio_block_f32_t *block) +{ + uint32_t mask = (0x80000000 >> (31 - (block->memory_pool_index & 0x1F))); + uint32_t index = block->memory_pool_index >> 5; + + __disable_irq(); + if (block->ref_count > 1) { + block->ref_count--; + } else { + //Serial.print("release_f32:"); + //Serial.println((uint32_t)block, HEX); + f32_memory_pool_available_mask[index] |= mask; + f32_memory_used--; + } + __enable_irq(); +} + +// Transmit an audio data block +// to all streams that connect to an output. The block +// becomes owned by all the recepients, but also is still +// owned by this object. Normally, a block must be released +// by the caller after it's transmitted. This allows the +// caller to transmit to same block to more than 1 output, +// and then release it once after all transmit calls. +void AudioStream_F32::transmit(audio_block_f32_t *block, unsigned char index) +{ + for (AudioConnection_F32 *c = destination_list_f32; c != NULL; c = c->next_dest) { + if (c->src_index == index) { + if (c->dst.inputQueue_f32[c->dest_index] == NULL) { + c->dst.inputQueue_f32[c->dest_index] = block; + block->ref_count++; + } + } + } +} + +// Receive block from an input. The block's data +// may be shared with other streams, so it must not be written +audio_block_f32_t * AudioStream_F32::receiveReadOnly_f32(unsigned int index) +{ + audio_block_f32_t *in; + + if (index >= num_inputs_f32) return NULL; + in = inputQueue_f32[index]; + inputQueue_f32[index] = NULL; + return in; +} + + +// Receive block from an input. The block will not +// be shared, so its contents may be changed. +audio_block_f32_t * AudioStream_F32::receiveWritable_f32(unsigned int index) +{ + audio_block_f32_t *in, *p; + + if (index >= num_inputs_f32) return NULL; + in = inputQueue_f32[index]; + inputQueue_f32[index] = NULL; + if (in && in->ref_count > 1) { + p = allocate_f32(); + if (p) memcpy(p->data, in->data, sizeof(p->data)); + in->ref_count--; + in = p; + } + return in; +} + +void AudioConnection_F32::connect(void) { + AudioConnection_F32 *p; + + if (dest_index > dst.num_inputs_f32) return; + __disable_irq(); + p = src.destination_list_f32; + if (p == NULL) { + src.destination_list_f32 = this; + } else { + while (p->next_dest) p = p->next_dest; + p->next_dest = this; + } + src.active = true; + dst.active = true; + __enable_irq(); +} diff --git a/AudioStream_F32.h b/AudioStream_F32.h new file mode 100644 index 0000000..8fbee44 --- /dev/null +++ b/AudioStream_F32.h @@ -0,0 +1,98 @@ +/* + * AudioStream_F32 + * + * Created: Chip Audette, November 2016 + * Purpose; Extend the Teensy Audio Library's "AudioStream" to permit floating-point audio data. + * + * I modeled it directly on the Teensy code in "AudioStream.h" and "AudioStream.cpp", which are + * available here: https://github.com/PaulStoffregen/cores/tree/master/teensy3 + * + * MIT License. use at your own risk. +*/ + +#ifndef _OpenAudio_ArduinoLibrary +#define _OpenAudio_ArduinoLibrary + +#include //ARM DSP extensions. for speed! +#include //Teensy Audio Library + +class AudioStream_F32; +class AudioConnection_F32; + +//create a new structure to hold audio as floating point values. +//modeled on the existing teensy audio block struct, which uses Int16 +//https://github.com/PaulStoffregen/cores/blob/268848cdb0121f26b7ef6b82b4fb54abbe465427/teensy3/AudioStream.h +typedef struct audio_block_f32_struct { + unsigned char ref_count; + unsigned char memory_pool_index; + unsigned char reserved1; + unsigned char reserved2; + float data[AUDIO_BLOCK_SAMPLES]; // AUDIO_BLOCK_SAMPLES is 128, from AudioStream.h + const int length = AUDIO_BLOCK_SAMPLES; // AUDIO_BLOCK_SAMPLES is 128, from AudioStream.h + const float fs_Hz = AUDIO_SAMPLE_RATE; // AUDIO_SAMPLE_RATE is 44117.64706 from AudioStream.h +} audio_block_f32_t; + +class AudioConnection_F32 +{ + public: + AudioConnection_F32(AudioStream_F32 &source, AudioStream_F32 &destination) : + src(source), dst(destination), src_index(0), dest_index(0), + next_dest(NULL) + { connect(); } + AudioConnection_F32(AudioStream_F32 &source, unsigned char sourceOutput, + AudioStream_F32 &destination, unsigned char destinationInput) : + src(source), dst(destination), + src_index(sourceOutput), dest_index(destinationInput), + next_dest(NULL) + { connect(); } + friend class AudioStream_F32; + protected: + void connect(void); + AudioStream_F32 &src; + AudioStream_F32 &dst; + unsigned char src_index; + unsigned char dest_index; + AudioConnection_F32 *next_dest; +}; + +#define AudioMemory_F32(num) ({ \ + static audio_block_f32_t data[num]; \ + AudioStream_F32::initialize_f32_memory(data, num); \ +}) + +class AudioStream_F32 : public AudioStream { + public: + AudioStream_F32(unsigned char n_input_f32, audio_block_f32_t **iqueue) : AudioStream(1, inputQueueArray_i16), + num_inputs_f32(n_input_f32), inputQueue_f32(iqueue) { + //active_f32 = false; + destination_list_f32 = NULL; + for (int i=0; i < n_input_f32; i++) { + inputQueue_f32[i] = NULL; + } + }; + static void initialize_f32_memory(audio_block_f32_t *data, unsigned int num); + //virtual void update(audio_block_f32_t *) = 0; + static uint8_t f32_memory_used; + static uint8_t f32_memory_used_max; + + protected: + //bool active_f32; + unsigned char num_inputs_f32; + static audio_block_f32_t * allocate_f32(void); + static void release(audio_block_f32_t * block); + void transmit(audio_block_f32_t *block, unsigned char index = 0); + audio_block_f32_t * receiveReadOnly_f32(unsigned int index = 0); + audio_block_f32_t * receiveWritable_f32(unsigned int index = 0); + friend class AudioConnection_F32; + private: + AudioConnection_F32 *destination_list_f32; + audio_block_f32_t **inputQueue_f32; + virtual void update(void) = 0; + audio_block_t *inputQueueArray_i16[1]; //two for stereo + static audio_block_f32_t *f32_memory_pool; + static uint32_t f32_memory_pool_available_mask[6]; +}; + + + +#endif \ No newline at end of file diff --git a/OpenAudio_ArduinoLibrary.h b/OpenAudio_ArduinoLibrary.h new file mode 100644 index 0000000..b7f7286 --- /dev/null +++ b/OpenAudio_ArduinoLibrary.h @@ -0,0 +1,4 @@ + +#include +#include +#include diff --git a/keywords.txt b/keywords.txt new file mode 100644 index 0000000..e69de29 diff --git a/readme.md b/readme.md new file mode 100644 index 0000000..e69de29