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Encoding and Decoding of Audio
Unfortunately the available memory on Microcontrollers is quite restricted and we do not get very far by storing a (uncompressed) WAV file e.g. in program/flesh memory, so I started to look into compressed audio formats: The compression and decompression can be done with the help of Codecs. Codecs are also important if you need to transmit audio data at a high sampling rate over a line with limited capacity.
On the desktop we can use the FFmpeg project which comes with a rich set of functionality. Unfortunately the situation is much more fragmented for Microcontrollers.
I started to collect the relevant libraries and in order to make things simple to use I also added a simple C++ API on top of the available libraries:
Project | Class | Include | Function | Format |
---|---|---|---|---|
- | Decoder8Bit | AudioCodecs/Codec8Bit.h | Decoding | PCM |
- | Encoder8Bit | AudioCodecs/Codec8Bit.h | Encoding | PCM |
- | WAVDecoder | AudioCodecs/CodecWAV.h | Decoding | WAV |
- | WAVEncoder | AudioCodecs/CodecWAV.h | Encoding | WAV |
libhelix | MP3DecoderHelix | AudioCodecs/CodecMP3Helix.h | Decoding | MP3 |
libhelix | AACDecoderHelix | AudioCodecs/CodecAACHelix.h | Decoding | AAC |
libvorbis-idec | VorbisDecoder | AudioCodecs/CodecVorbis.h | Decoding | OGG Vorbis |
fdk-aac | AACDecoderFDK | AudioCodecs/CodecAACFDK.h | Encoding, Decoding | AAC |
libmad | MP3DecoderMAD | AudioCodecs/CodecMP3MAD.h | Decoding | MP3 |
liblame | MP3EncoderLAME | AudioCodecs/CodecMP3LAME.h | Encoding | MP3 |
libflac | FLACDecoder | AudioCodecs/CodecFLAC.h | Decoding | FLAC |
libflac | FLACEncoder | AudioCodecs/CodecFLAC.h | Encoding | FLAC |
minimp3 | MP3DecoderMini | AudioCodecs/CodecMP3Mini.h | Decoding | MP3 |
libsbc | SBCDecoder | AudioCodecs/CodecSBC.h | Decoding | SBC |
libsbc | SBCEncoder | AudioCodecs/CodecSBC.h | Encoding | SBC |
liblc3 | LC3Decoder | AudioCodecs/CodecLC3.h | Decoding | LC3 |
liblc3 | LC3Encoder | AudioCodecs/CodecLC3.h | Encoding | LC3 |
libopenaptx | APTXDecoder | AudioCodecs/CodecAPTX.h | Decoding | APTX |
libopenaptx | APTXEncoder | AudioCodecs/CodecAPTX.h | Encoding | APTX |
libopus | OpusDecoder | AudioCodecs/CodecOpus.h | Decoding | Opus |
libopus | OpusEncoder | AudioCodecs/CodecOpus.h | Encoding | Opus |
libgsm | GSMDecoder | AudioCodecs/CodecGSM.h | Decoding | GSM |
libgsm | GSMEncoder | AudioCodecs/CodecGSM.h | Encoding | GSM |
libg722 | G722Decoder | AudioCodecs/CodecG722.h | Decoding | G.722 |
libg722 | G722Encoder | AudioCodecs/CodecG722.h | Encoding | G.722 |
codec2 | Codec2Decoder | AudioCodecs/CodecCodec2.h | Decoding | Codec2 |
codec2 | Codec2Encoder | AudioCodecs/CodecCodec2.h | Encoding | Codec2 |
I am also providing an integration into my Arduino Audio Tools where you can use these libraries with the EncodedAudioStream class:
#include "Arduino.h"
#include "AudioTools.h"
#include "AudioCodecs/CodecMP3Helix.h"
#include "BabyElephantWalk60_mp3.h"
MemoryStream mp3(BabyElephantWalk60_mp3, BabyElephantWalk60_mp3_len); // MP3 data source
I2SStream i2s; // final output of decoded stream
EncodedAudioStream dec(&i2s, new MP3DecoderHelix()); // Decoding stream
StreamCopy copier(dec, mp3); // copy in to out
void setup(){
Serial.begin(115200);
i2s.begin();
dec.begin();
}
void loop(){
if (mp3) {
copier.copy();
}
}
The above stream is implementing the following flow: mp3 MemoryStream -copy-> EncodedAudioStream -> I2SStream This method is used by most codecs.
Ogg and Vorbis are using an alternative methods which is pulling the data directly from an input stream:
#include "AudioTools.h"
#include "AudioCodecs/CodecVorbis.h"
const char* ssid = "ssid";
const char* pwd = "password";
URLStream url(ssid, pwd);
VorbisDecoder dec;
I2SStream i2s;
void setup() {
Serial.begin(115200);
AudioLogger::instance().begin(Serial, AudioLogger::Info);
i2s.begin(i2s.defaultConfig(TX_MODE));
url.begin("http://marmalade.scenesat.com:8086/bitjam.ogg","application/ogg");
dec.setInputStream(url);
dec.setOutputStream(i2s);
dec.begin();
}
void loop() {
dec.copy();
}
The encoding of audio data to a different format is also done with the help of the EncodedAudioStream class. The only difference (to the decoding examples) is that we pass an Encoder as argument.
Here is the related Arduino sketch:
#include "AudioTools.h"
#include "SdFat.h"
uint16_t sample_rate = 44100;
uint8_t channels = 2; // The stream will have 2 channels
NoiseGenerator<int16_t> noise(32000); // subclass of SoundGenerator with max amplitude of 32000
GeneratedSoundStream<int16_t> in_stream(noise); // Stream generated from sine wave
SdFat SD;
File audioFile; // final output stream
EncodedAudioStream out_stream(&audioFile, new WAVEncoder()); // encode as wav file
StreamCopy copier(out_stream, in_stream); // copies sound to out
void setup(){
Serial.begin(115200);
AudioLogger::instance().begin(Serial, AudioLogger::Info);
auto cfg = noise.defaultConfig();
cfg.sample_rate = sample_rate;
cfg.channels = channels;
cfg.bits_per_sample = 16;
noise.begin(cfg);
in_stream.begin();
// we need to provide the audio information to the encoder
out_stream.begin(cfg);
// open the output file
SD.begin(SdSpiConfig(PIN_CS, DEDICATED_SPI, SD_SCK_MHZ(2)));
audioFile = SD.open("/test/002.wav", O_WRITE | O_CREAT);
}
void loop(){
copier.copy();
}
We create an input stream which is based on some sound generator. In the out_stream we indicate the final output stream (which is a file in the example) and the encoder that is used when the data is written: GeneratedSoundStream -copy-> EncodedAudioStream -> File