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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:
- AudioDecoder convert an encoded format to 16 bit PCM
- AudioEncoder convert from 16 bit PCM data to the encoded format.
Library | Class | Include | Function | Format | Application |
---|---|---|---|---|---|
- | DecoderL8 | AudioCodecs/CodecL8.h | Decoding | PCM | Audio |
- | EncoderL8 | AudioCodecs/CodecL8.h | Encoding | PCM | Audio |
- | DecoderL16 | AudioCodecs/CodecL16.h | Decoding | PCM | Audio |
- | EncoderL16 | AudioCodecs/CodecL16.h | Encoding | PCM | Audio |
- | DecoderFloat | AudioCodecs/CodecFloat.h | Decoding | PCM | Audio |
- | EncoderFloat | AudioCodecs/CodecFloat.h | Encoding | PCM | Audio |
- | WAVDecoder | AudioCodecs/CodecWAV.h | Decoding | WAV | Audio |
- | WAVEncoder | AudioCodecs/CodecWAV.h | Encoding | WAV | Audio |
- | WavIMADecoder | AudioCodecs/CodecWavIMA.h | Decoding | WAV/IMA | Audio |
- | DecoderBase64 | AudioCodecs/CodecBase64.h | Decoding | Base64 | |
- | ADTSDecoder | AudioCodecs/CodecADTS.h | Decoding | ADTS | Audio |
- | EncoderBase64 | AudioCodecs/CodecBase64.h | Encoding | Base64 | |
libhelix | MP3DecoderHelix | AudioCodecs/CodecMP3Helix.h | Decoding | MP3 | Audio |
libmad | MP3DecoderMAD | AudioCodecs/CodecMP3MAD.h | Decoding | MP3 | Audio |
minimp3 | MP3DecoderMini | AudioCodecs/CodecMP3Mini.h | Decoding | MP3 | Audio |
liblame | MP3EncoderLAME | AudioCodecs/CodecMP3LAME.h | Encoding | MP3 | Audio |
libhelix | AACDecoderHelix | AudioCodecs/CodecAACHelix.h | Decoding | AAC | Audio |
libfaad | AACDecoderFAAD | AudioCodecs/CodecAACFAAD.h | Decoding | AAC | Audio |
fdk-aac | AACDecoderFDK | AudioCodecs/CodecAACFDK.h | Decoding | AAC | Audio |
fdk-aac | AACEncoderFDK | AudioCodecs/CodecAACFDK.h | Encoding | AAC | Audio |
libflac | FLACDecoder | AudioCodecs/CodecFLAC.h | Decoding | FLAC | Audio |
libflac | FLACEncoder | AudioCodecs/CodecFLAC.h | Encoding | FLAC | Audio |
libfoxenflac | FLACDecoderFoxen | AudioCodecs/CodecFLACFoxen.h | Decoding | FLAC | Audio |
libvorbis-tremor | VorbisDecoder | AudioCodecs/CodecVorbis.h | Decoding | OGG Vorbis | Audio |
libsbc | SBCDecoder | AudioCodecs/CodecSBC.h | Decoding | SBC | Audio |
libsbc | SBCEncoder | AudioCodecs/CodecSBC.h | Encoding | SBC | Audio |
liblc3 | LC3Decoder | AudioCodecs/CodecLC3.h | Decoding | LC3 | Audio |
liblc3 | LC3Encoder | AudioCodecs/CodecLC3.h | Encoding | LC3 | Audio |
libopenaptx | APTXDecoder | AudioCodecs/CodecAPTX.h | Decoding | APTX | Audio |
libopenaptx | APTXEncoder | AudioCodecs/CodecAPTX.h | Encoding | APTX | Audio |
libopus | OpusAudioDecoder | AudioCodecs/CodecOpus.h | Decoding | Opus | Audio |
libopus | OpusAudioEncoder | AudioCodecs/CodecOpus.h | Encoding | Opus | Audio |
libopus | OpusOggDecoder | AudioCodecs/CodecOpusOgg.h | Decoding | Opus | Audio |
libopus | OpusOggEncoder | AudioCodecs/CodecOpusOgg.h | Encoding | Opus | Audio |
adpcm | ADPCMDecoder | AudioCodecs/CodecADPCM.h | Decoding | ADPCM | Audio |
adpcm | ADPCMEncoder | AudioCodecs/CodecADPCM.h | Encoding | ADPCM | Audio |
adpcm-xq | ADPCMDecoderXQ | AudioCodecs/CodecADPCM.h | Decoding | ADPCM | Audio |
adpcm-xq | ADPCMEncoderXQ | AudioCodecs/CodecADPCM.h | Encoding | ADPCM | Audio |
libgsm | GSMDecoder | AudioCodecs/CodecGSM.h | Decoding | GSM | Speech |
libgsm | GSMEncoder | AudioCodecs/CodecGSM.h | Encoding | GSM | Speech |
libg7xx | G711_ALAWDecoder | AudioCodecs/CodecG7xx.h | Decoding | ALAW | Speech |
libg7xx | G711_ALAWEncoder | AudioCodecs/CodecG7xx.h | Encoding | ALAW | Speech |
libg7xx | G711_ULAWDecoder | AudioCodecs/CodecG7xx.h | Decoding | ULAW | Speech |
libg7xx | G711_ULAWEncoder | AudioCodecs/CodecG7xx.h | Encoding | ULAW | Speech |
libg7xx | G721Decoder | AudioCodecs/CodecG7xx.h | Decoding | G.721 | Speech |
libg7xx | G721Encoder | AudioCodecs/CodecG7xx.h | Encoding | G.721 | Speech |
libg722 | G722Decoder | AudioCodecs/CodecG722.h | Decoding | G.722 | Speech |
libg722 | G722Encoder | AudioCodecs/CodecG722.h | Encoding | G.722 | Speech |
libg7xx | G723_24Decoder | AudioCodecs/CodecG7xx.h | Decoding | G.723 | Speech |
libg7xx | G723_24Encoder | AudioCodecs/CodecG7xx.h | Encoding | G.723 | Speech |
libg7xx | G723_40Decoder | AudioCodecs/CodecG7xx.h | Decoding | G.723 | Speech |
libg7xx | G723_40Encoder | AudioCodecs/CodecG7xx.h | Encoding | G.723 | Speech |
codec2 | Codec2Decoder | AudioCodecs/CodecCodec2.h | Decoding | Codec2 | Speech |
codec2 | Codec2Encoder | AudioCodecs/CodecCodec2.h | Encoding | Codec2 | Speech |
Library | Class | Include | Function | Format | Application |
---|---|---|---|---|---|
libopus | OggContainerDecoder | AudioCodecs/ContainerOgg.h | Decoding | Ogg | Audio |
libopus | OggContainerEncoder | AudioCodecs/ContainerOgg.h | Encoding | Ogg | Audio |
- | BinaryContainerEncoder | AudioCodecs/ContainerBinary.h | Encoding | - | Audio |
- | BinaryContainerDecoder | AudioCodecs/ContainerBinary.h | Decoding | - | Audio |
- | AVIDecoder | AudioCodecs/ContainerAVI.h | Decoding | AVI | Video |
- | MTSDecoder | AudioCodecs/CodecMTS.h | Decoding | MPEG-TS (MTS) | Audio |
tsdemux | MTSDecoder | AudioCodecs/CodecMTS.h | Decoding | video/mp2t MPEG-TS MTS | Audio Video |
If you want to use a codec, do not forget that you need to install the related library!
Most decoders inherit from AudioDecoder. I am also providing an integration into my Arduino Audio Tools where you can use these libraries with the EncodedAudioStream class:
#include "AudioTools.h"
#include "AudioTools/AudioCodecs/CodecMP3Helix.h"
#include "BabyElephantWalk60_mp3.h"
MemoryStream data(BabyElephantWalk60_mp3, BabyElephantWalk60_mp3_len); // MP3 data source
I2SStream i2s; // final output of decoded stream
MP3DecoderHelix mp3; // Codec
EncodedAudioStream dec(&i2s, &mp3); // Decoding stream
StreamCopy copier(dec, data); // 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.
I have added the MultiDecoder class to support the decoding of multiple data formats. The actual decoder is only opened at it's first use: The relevant decoder is determined dynamically at the first write from the determined mime type. You can add your own custom mime type determination logic.
When you change the data source to provide a new format you need to call end() on the EncodedAudioStream, to let the decoder know that it needs to determine a new format.
#include "AudioTools.h"
#include "AudioTools/AudioCodecs/CodecMP3Helix.h"
#include "AudioTools/AudioCodecs/CodecAACHelix.h"
#include "AudioTools/AudioLibs/AudioBoardStream.h"
URLStream url("ssid","password");
I2SStream i2s;
MultiDecoder multi;
MP3DecoderHelix mp3;
AACDecoderHelix aac;
WAVDecoder wav;
EncodedAudioStream dec(&i2s, &multi); // Decoding stream
StreamCopy copier(dec, url); // copy url to decoder
void setup(){
Serial.begin(115200);
AudioToolsLogger.begin(Serial, AudioToolsLogLevel::Info);
// register supported codecs with their mime type
multi.addDecoder(mp3, "audio/mpeg");
multi.addDecoder(aac, "audio/aac");
multi.addDecoder(wav, "audio/vnd.wave");
// setup i2s
auto config = i2s.defaultConfig(TX_MODE);
// you could define e.g your pins and change other settings
//config.pin_ws = 10;
//config.pin_bck = 11;
//config.pin_data = 12;
//config.mode = I2S_STD_FORMAT;
i2s.begin(config);
// setup I2S based on sampling rate provided by decoder
dec.begin();
// mp3 radio
url.begin("http://stream.srg-ssr.ch/m/rsj/mp3_128","audio/mpeg");
}
void loop(){
copier.copy();
}
Please note:
- The distingtion between mp3 and aac just from their content is difficult and can't be 100% reliable. Files are more reliable because mp3 usually starts with some metadata.
- You can provide the URLStream object in the constructor of the MultiDecoder, so that the system can look up the mime type from the http response.
- The MimeDetector contains different alternatives predefined implementations that you can use when you add a decoder.
- If you don't want to rely on the automatic determination you can select the decoder by calling the selectDecoder method.
You can also decode on the input side, which is less efficient, but sometimes more convenient. This is implementing the following flow: mp3 MemoryStream -> EncodedAudioStream -copy-> I2SStream
#include "AudioTools.h"
#include "AudioTools/AudioCodecs/CodecMP3Helix.h"
#include "BabyElephantWalk60_mp3.h"
MemoryStream data(BabyElephantWalk60_mp3, BabyElephantWalk60_mp3_len); // MP3 data source
MP3DecoderHelix mp3;
EncodedAudioStream dec(&data, &mp3); // Decoding stream
I2SStream i2s; // final output of decoded stream
StreamCopy copier(i2s, dec); // copy dec to out
void setup(){
Serial.begin(115200);
i2s.begin();
dec.addNotifyAudioChange(i2s);
//dec.resizeReadResultQueue(1024 * 10);
dec.begin();
}
void loop(){
copier.copy();
}
- Please note that the read functionality is implemented by reading the data from the indicated source and calling the write functionality on the decoder to store the result in a queue. Calling readBytes() is then providing the data from this queue.
- The MP3 decoder provides the PCM data in big arrays (e.g. 4608 bytes). The default decoding queue might not be big enough to store this result, so you might need to specify the queue size yourself. You can do this by setting it to a fixed size by calling dec.resizeReadResultQueue(1024 * 10); .
Because of this complexity I do not recommend this functionality for beginners and only use it if you have a good understanding of the result sizes provided by the involved functionality.
Ogg (FLACDecoder) and Vorbis VorbisDecoder are using an alternative method which is pulling the data directly from an input stream:
In this case a StreamingDecoder is used.
#include "AudioTools.h"
#include "AudioTools/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");
// setup decoder
dec.setInputStream(url);
dec.setOutputStream(i2s);
dec.begin();
}
void loop() {
dec.copy();
}
You can transform any AudioDecoder into a StreamingDecoder with the help of a StreamingDecoderAdapter.
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
WAVEncoder encoder;
EncodedAudioStream out_stream(&audioFile, &encoder); // 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();
// audioFile.flush(); // force write down of data
}
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
Please note the following:
- you need to make sure that the file content is written to the file by calling audioFile.close() at the end - or by flushing the individual writes.
- call out_stream.begin() before you write to a new file, this makes sure that the header is written to the file if necessary (e.g. for WAV files).
- Before you start to write to a file, delete the file or move to the beginning. Otherwise the content is just appended!
- This example is using the sdfat library, but you can use any other Arduino file library implementation.
- MP3 and AAC are quire popular audio format, but they require a lot of memory and are at the edge what a Microcontroller can do. I recommend to avoid them and to prefer a lean format like ADPCM.