Files
klubhaus-doorbell/libraries/FastLED/src/fl/audio_input.h
2026-02-12 00:45:31 -08:00

143 lines
5.2 KiB
C++

#pragma once
#pragma once
#include "fl/stdint.h"
#include "fl/int.h"
#include "fl/vector.h"
#include "fl/variant.h"
#include "fl/shared_ptr.h"
#include "fl/audio.h"
#include "fl/compiler_control.h"
#include "platforms/audio.h"
#ifndef FASTLED_HAS_AUDIO_INPUT
#error "platforms/audio.h must define FASTLED_HAS_AUDIO_INPUT"
#endif
#define I2S_AUDIO_BUFFER_LEN 512
#define AUDIO_DEFAULT_SAMPLE_RATE 44100ul
#define AUDIO_DEFAULT_BIT_RESOLUTION 16
#define AUDIO_DMA_BUFFER_COUNT 8
namespace fl {
// Note: Right now these are esp specific, but they are designed to migrate to a common api.
enum AudioChannel {
Left = 0,
Right = 1,
Both = 2, // Two microphones can be used to capture both channels with one AudioSource.
};
enum I2SCommFormat {
Philips = 0X01, // I2S communication I2S Philips standard, data launch at second BCK
MSB = 0X02, // I2S communication MSB alignment standard, data launch at first BCK
PCMShort = 0x04, // PCM Short standard, also known as DSP mode. The period of synchronization signal (WS) is 1 bck cycle.
PCMLong = 0x0C, // PCM Long standard. The period of synchronization signal (WS) is channel_bit*bck cycles.
Max = 0x0F, // standard max
};
struct AudioConfigI2S {
int mPinWs;
int mPinSd;
int mPinClk;
int mI2sNum;
AudioChannel mAudioChannel;
u16 mSampleRate;
u8 mBitResolution;
I2SCommFormat mCommFormat;
bool mInvert;
AudioConfigI2S(
int pin_ws,
int pin_sd,
int pin_clk,
int i2s_num,
AudioChannel mic_channel,
u16 sample_rate,
u8 bit_resolution,
I2SCommFormat comm_format = Philips,
bool invert = false
)
: mPinWs(pin_ws), mPinSd(pin_sd), mPinClk(pin_clk),
mI2sNum(i2s_num), mAudioChannel(mic_channel),
mSampleRate(sample_rate), mBitResolution(bit_resolution), mCommFormat(comm_format), mInvert(invert) {}
};
struct AudioConfigPdm {
int mPinDin;
int mPinClk;
int mI2sNum;
u16 mSampleRate;
bool mInvert = false;
AudioConfigPdm(int pin_din, int pin_clk, int i2s_num, u16 sample_rate = AUDIO_DEFAULT_SAMPLE_RATE, bool invert = false)
: mPinDin(pin_din), mPinClk(pin_clk), mI2sNum(i2s_num), mSampleRate(sample_rate), mInvert(invert) {}
};
class AudioConfig : public fl::Variant<AudioConfigI2S, AudioConfigPdm> {
public:
// The most common microphone on Amazon as of 2025-September.
static AudioConfig CreateInmp441(int pin_ws, int pin_sd, int pin_clk, AudioChannel channel, u16 sample_rate = 44100ul, int i2s_num = 0) {
AudioConfigI2S config(pin_ws, pin_sd, pin_clk, i2s_num, channel, sample_rate, 16);
return AudioConfig(config);
}
AudioConfig(const AudioConfigI2S& config) : fl::Variant<AudioConfigI2S, AudioConfigPdm>(config) {}
AudioConfig(const AudioConfigPdm& config) : fl::Variant<AudioConfigI2S, AudioConfigPdm>(config) {}
};
class IAudioInput {
public:
// This is the single factory function for creating the audio source. If the creation was successful, then
// the return value will be non-null. If the creation was not successful, then the return value will be null
// and the error_message will be set to a non-empty string.
// Keep in mind that the AudioConfig is a variant type. Many esp types do not support all the types in the variant.
// For example, the AudioConfigPdm is not supported on the ESP32-C3 and in this case it will return a null pointer
// and the error_message will be set to a non-empty string.
// Implimentation notes:
// It's very important that the implimentation uses a esp task / interrupt to fill in the buffer. The reason is that
// there will be looooong delays during FastLED show() on some esp platforms, for example idf 4.4. If we do
// poll only, then audio buffers can be dropped. However if using a task then the audio buffers will be
// set internally via an interrupt / queue and then they can just be popped off the queue.
static fl::shared_ptr<IAudioInput> create(const AudioConfig& config, fl::string* error_message = nullptr);
virtual ~IAudioInput() = default;
// Starts the audio source.
virtual void start() = 0;
// Stops the audio source, call this before light sleep.
virtual void stop() = 0;
virtual bool error(fl::string* msg = nullptr) = 0; // if an error occured then query it here.
// Read audio data and return as AudioSample with calculated timestamp.
// Returns invalid AudioSample on error or when no data is available.
virtual AudioSample read() = 0;
// Read all available audio data and return as AudioSample. All AudioSamples
// returned by this will be valid.
size_t readAll(fl::vector_inlined<AudioSample, 16> *out) {
size_t count = 0;
while (true) {
AudioSample sample = read();
if (sample.isValid()) {
out->push_back(sample);
count++;
} else {
break;
}
}
return count;
}
};
// Free function for audio input creation - can be overridden by platform-specific implementations
fl::shared_ptr<IAudioInput> platform_create_audio_input(const AudioConfig& config, fl::string* error_message = nullptr) FL_LINK_WEAK;
} // namespace fl