How to increase the microphone volume when using streams-i2s-webserver_wav

[TheAxial]

Hello. please tell me how to increase the output volume of the i2s microphone?
I tried using
ConverterScaler<int16_t> volume(1.0, 0, 32767);
And
volume.setFactor(100.0); (substituted values from 0.5 to 100.0)
Here is the full code:
`/**

• @file streams-i2s-webserver_wav.ino
• This sketch reads sound data from I2S. The result is provided as WAV stream which can be listened to in a Web Browser
• @author Phil Schatzmann
• @copyright GPLv3
  */

#include "AudioTools.h"
using namespace audio_tools;

AudioWAVServer server("ssid","pass"); // the same a above

I2SStream i2sStream; // Access I2S as stream
ConverterFillLeftAndRight<int16_t> filler(LeftIsEmpty); // fill both channels - or change to RightIsEmpty
ConverterScaler<int16_t> volume(1.0, 0, 32767);

void setup(){
Serial.begin(115200);
AudioLogger::instance().begin(Serial, AudioLogger::Info);
volume.setFactor(100.0);
Serial.println("starting I2S...");
auto config = i2sStream.defaultConfig(RX_MODE); //I2S_STD_FORMAT, I2S_MSB_FORMAT, I2S_PHILIPS_FORMAT
config.i2s_format = I2S_PCM_SHORT; // if quality is bad change to I2S_LSB_FORMAT #23
config.sample_rate = 22050;
config.channels = 1;
config.bits_per_sample = 32;
i2sStream.begin(config);
Serial.println("I2S started");

// start data sink
server.begin(i2sStream, config, &filler);
}

// Arduino loop
void loop() {
// Handle new connections
server.doLoop();
}`

[pschatzmann]

With a factor from 0.0 to 1.0 the values are decreased.
With a factor > 1.0 the values are amplified: Typical values might be 1.5 or 2.0 (which will double the values)
I think a factor of 100.0 is just crazy and will cause that most values are clipped.

If you get the values from I2S, they are oscillating between 0, so the scaler would need to be
ConverterScaler<int16_t> volume(1.0, -32768, 32767);

[TheAxial]

I'm sorry, I accidentally
The volume is not adjustable at all. Even if I install volume.setFactor(0.0);
The sound will still be the same as without this line

[pschatzmann]

That's because you do not use volume anywhere: You will need to pass it as argument to the server.begin(). Since you want to use multiple converters you will need to wrap them in a MultiConverter. Maybe just to test this first replace the &filler argument with &volume

[TheAxial]

No changes. I replaced the &filler argument with &volume, it didn't help. Changed the values in ConverterScaler<int16_t> volume(1.0, -32768, 32767); to completely different ranges from -1023..1023 to -10000000..10000000. Changed volume.setFactor(5.0);, tried 1.5, 2.0, 5.0, 20.0. Reactions 0. Please help me

[pschatzmann]

Oh, I just noticed that you have config.bits_per_sample = 32; so this changes everything and you need to ajust the range values! -2147483648l to 2147483647l
Dont forget to change the definition of the filters as well: e.g. ConverterFillLeftAndRight<int32_t> and ConverterScaler<int32_t>.

16 bits -> int16_t
32 bits -> int32_t

[TheAxial]

An error pops up as soon as I change to 32

C:\Users\TheAxial\Documents\Arduino\streams-i2s-webserver_wav\streams-i2s-webserver_wav.ino: In function 'void setup()':
streams-i2s-webserver_wav:21:42: error: no matching function for call to 'audio_tools::AudioWAVServer::begin(audio_tools::I2SStream&, audio_tools::I2SConfig&, audio_tools::ConverterScaler)'
server.begin(i2sStream, config, &volume);
^
In file included from C:\Users\TheAxial\Documents\Arduino\libraries\arduino-audio-tools-main\src/AudioHttp/AudioHttp.h:4,
from C:\Users\TheAxial\Documents\Arduino\libraries\arduino-audio-tools-main\src/AudioTools.h:30,
from C:\Users\TheAxial\Documents\Arduino\streams-i2s-webserver_wav\streams-i2s-webserver_wav.ino:1:
C:\Users\TheAxial\Documents\Arduino\libraries\arduino-audio-tools-main\src/AudioHttp/AudioServer.h:271:14: note: candidate: 'void audio_tools::AudioEncoderServer::begin(Stream&, int, int, int, audio_tools::BaseConverter)'
void begin(Stream &in, int sample_rate, int channels, int bits_per_sample=16, BaseConverter<int16_t> converter=nullptr) {
^~~~~
C:\Users\TheAxial\Documents\Arduino\libraries\arduino-audio-tools-main\src/AudioHttp/AudioServer.h:271:14: note: no known conversion for argument 2 from 'audio_tools::I2SConfig' to 'int'
C:\Users\TheAxial\Documents\Arduino\libraries\arduino-audio-tools-main\src/AudioHttp/AudioServer.h:289:14: note: candidate: 'void audio_tools::AudioEncoderServer::begin(Stream&, audio_tools::AudioBaseInfo, audio_tools::BaseConverter)'
void begin(Stream &in, AudioBaseInfo info, BaseConverter<int16_t> converter=nullptr) {
^~~~~
C:\Users\TheAxial\Documents\Arduino\libraries\arduino-audio-tools-main\src/AudioHttp/AudioServer.h:289:14: note: no known conversion for argument 3 from 'audio_tools::ConverterScaler' to 'audio_tools::BaseConverter'
C:\Users\TheAxial\Documents\Arduino\libraries\arduino-audio-tools-main\src/AudioHttp/AudioServer.h:306:14: note: candidate: 'void audio_tools::AudioEncoderServer::begin(audio_tools::AudioServerDataCallback, int, int, int)'
void begin(AudioServerDataCallback cb, int sample_rate, int channels, int bits_per_sample=16) {
^~~~~
C:\Users\TheAxial\Documents\Arduino\libraries\arduino-audio-tools-main\src/AudioHttp/AudioServer.h:306:14: note: no known conversion for argument 1 from 'audio_tools::I2SStream' to 'audio_tools::AudioServerDataCallback' {aka 'void ()(Print*)'}
exit status 1
no matching function for call to 'audio_tools::AudioWAVServer::begin(audio_tools::I2SStream&, audio_tools::I2SConfig&, audio_tools::ConverterScaler*)'

code:
#include "AudioTools.h"
using namespace audio_tools;

AudioWAVServer server("ssid", "pass");

I2SStream i2sStream;
ConverterScaler<int32_t> volume(1.0, -2147483648l, 2147483647l);

void setup() {
Serial.begin(115200);
AudioLogger::instance().begin(Serial, AudioLogger::Info);
volume.setFactor(5.0);
Serial.println("starting I2S...");
auto config = i2sStream.defaultConfig(RX_MODE);
config.i2s_format = I2S_PCM_SHORT;
config.sample_rate = 22050;
config.channels = 1;
config.bits_per_sample = 32;
i2sStream.begin(config);
Serial.println("I2S started");
server.begin(i2sStream, config, &volume);
}

void loop() {
server.doLoop();
}

[pschatzmann]

OK, it seem that AudioServer only supports 16bits. I suggest to move back to 16 bits processing.
If you insist on using I2S with 24 bits, you will need to feed WebServer with a converted stream to 16bits.
E.g. by wrapping the I2SStream in a FormatConverterStream

[TheAxial]

I would be happy to stay on 16 bits, but I have a crackle in the speakers.
Microphone: INMP441

[TheAxial]

I'm not good at programming.
Could you please indicate where and what to write to me?
I'm making a radio babysitter

[pschatzmann]

I suggest that you forget about the converters. Read the chapter about volume control
In the config you need to indicate allow_boost = true to allow values > 1.0
Then you can pass the VolumeStream object to the WebServer begin.
This should work with any sample size!

[TheAxial]

If I understood you correctly, then I get an error that such a parameter cannot be passed

[pschatzmann]

That's for the configuration of the VolumeStream!

[pschatzmann]

VolumeStream volume(i2s);
...
in setup:

auto cfgVol = volume.defaultConfig();
cfgVol.allow_boost = true;
cfgVol.sample_rate = 22050;
cfgVol.channels = 1;
cfgVol.bits_per_sample = 32;
volume.begin(cfgVol);
volume.setVolume(1.3);

server.begin(volume, config);