M5 Atom Echo - i2s PDM to RTSP

[lfpoulain]

Hi everyone,

I'm currently working on a project with the M5 Atom Echo module, aiming to create an RTSP microphone for home automation and conversational agents. I've already explored the ESPHome examples, but I'm specifically interested in using RTSP for internal development.

Below is the current code I'm using:

#include "AudioTools.h"

AudioWAVServer server("SSID","MDP"); // the same as above

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

void setup(){
  Serial.begin(115200);
  AudioLogger::instance().begin(Serial, AudioLogger::Info);

  // start i2s input with default configuration
  Serial.println("starting I2S...");
  auto cfg = i2sStream.defaultConfig(RX_MODE);

  cfg.bits_per_sample = 16;
  cfg.channels = 1;
  cfg.sample_rate = 44100;
  cfg.signal_type = PDM;
  //cfg.channel_format = I2S_CHANNEL_FMT_ALL_LEFT;  // or left ?
  cfg.use_apll = false;
  cfg.auto_clear = false;
  cfg.pin_ws = 33;
  cfg.pin_data = 23;

  i2sStream.begin(cfg);
  Serial.println("I2S started");

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

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

The code almost works, but I'm encountering an issue where the sound is slowed down and takes a significant amount of time to load.

Any help or suggestions on how to resolve this issue would be greatly appreciated. Specifically, I'm looking for advice on the correct configuration for I2S PDM and any potential improvements to the RTSP streaming setup.

Thank you in advance for your assistance!

Best regards,
LFPoulain

[pschatzmann]

I think this is just the regular buffering done by your browser.
The use of the ConverterFillLeftAndRight does not make any sense if you have an mono signal as input and outpu!

[lfpoulain]

The issue with the slow and low-pitch voice was indeed "ConverterFillLeftAndRight"! Thank you.
To avoid browser issues, I am using VLC.
Now, I'm experiencing some glitches and cuts during listening.
How can I use RTSP instead of WAVServer? I looked into examples but I can't figure it out.
Maybe my case could end up as an i2s_pdm_rtsp example! 😆

[pschatzmann]

I think it is probably not worth the effort and you will not be able to handle such a high sample rate.
If you look at my example, I could only make it work the a sample rate of 16000

[pschatzmann]

If the sine example is working, I would not know why the microphone would not work...