Working example of URLStreamBuffered

[dired]

Dear Community and pschatzmann, thanks so much for this library!
I am used to Arduino IDE and whenever I try to use URLStreamBuffered instead of URLStream, I get "Compilation error: cannot declare variable 'urlStream' to be of abstract type 'audio_tools::URLStreamBuffered'".
In the Performance-Section of the wiki, it is said that UrlStreamBuffered is for plaforms that support FreeRTOS. I use the ESP32-A1s Audio kit, so the platform generally supports FreeRTOS. I have also "installed" the arduino-freertos-addons, but I wasn't able to use it properly so far, which also lead me to writing this question.

In general I want to stream from a "lossless" aac 96khz internet radio and it sounds good most of the time but each minute or so on average, I have those repeated samples/chunks (why does this happen at all, feels like it "hangs" shortly but I don't think it actually does, so why would it make this robotic noise instead of not repeating non-existent samples/chunks). I made use of all other options in the performance section and the best was more or less defaults with a little bit raised buffers on StreamCopy and i2s_cfg.buffer_size (probably cosmetically). But I really wanted to try and add network buffering and also maybe monitor when the buffer runs out and actually detect when the bottleneck is the wifi/network/(not so far existent)buffer.

P.S.: This is my now workaround-code (also wanted concurrency) after getting over copier so far (WIP, not nice and also maybe not better than copier), it sounds good most of the time as well (didn't came to measuring if it's less or more "crackle"/haning). But I still feel like missing out on UrlStreamBuffered ^^

#include "AudioTools.h"
#include "AudioCodecs/CodecAACHelix.h"
#include "AudioLibs/AudioBoardStream.h"
#include "Concurrency/SynchronizedBuffers.h"

const char *ssid = "wifisid";
const char *password = "wifipw";
const char *url = "https://motherearth.streamserver24.com/listen/motherearth_klassik/motherearth.klassik.aac";

#define bfsz 2048
#define bfcnt 512

URLStream urlStream(ssid, password);
AudioBoardStream i2s(AudioKitEs8388V1);
EncodedAudioStream decoder(&i2s, new AACDecoderHelix());
SynchronizedBufferRTOS<uint8_t> buffer(bfsz, bfcnt);

TaskHandle_t readerTaskHandle = NULL;
TaskHandle_t playerTaskHandle = NULL;

void readerTask(void *parameter) {
    uint8_t tempBuffer[512];
    while (true) {
        size_t bytesRead = urlStream.readBytes(tempBuffer, sizeof(tempBuffer));
        if (bytesRead > 0) {
            buffer.writeArray(tempBuffer, bytesRead);
        }
        vTaskDelay(1);
    }
}

void playerTask(void *parameter) {
    uint8_t tempBuffer[512];
    while (true) {
        size_t available = buffer.available();
        if (available >= sizeof(tempBuffer)) {
            size_t bytesRead = buffer.readArray(tempBuffer, sizeof(tempBuffer));
            decoder.write(tempBuffer, bytesRead);
        }
        vTaskDelay(1);
    }
}

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

    auto config = i2s.defaultConfig(TX_MODE);
    config.sample_rate = 96000;
    config.bits_per_sample = 16;
    config.channels = 2;
    config.buffer_size = 1024*2;
    i2s.begin(config);

    urlStream.begin(url, "audio/aac");

    decoder.begin();

    xTaskCreatePinnedToCore(readerTask, "ReaderTask", 10000, NULL, 1, &readerTaskHandle, 0);
    xTaskCreatePinnedToCore(playerTask, "PlayerTask", 10000, NULL, 1, &playerTaskHandle, 1);
}

void loop() {
    vTaskDelay(1000 / portTICK_PERIOD_MS);
}

[pschatzmann]

I committed a correction...

[dired]

It compiles fine now, thank you so much for the fast reply and action! I'm truly amazed by your drive and spirit!
I will think of some tests better than my subjective hearing now to check which software solution brings least errors and where the bottlenecks are, really curious about it all, and your libraries and articles were the awesomest start for using my esp32-a1s! This "radio"-code was just an initial idea because I wanted/needed to break-in some new headphones (because the manufacturer actually recommended that). I later wanted to measure and compare some DACs and maybe codecs. I wonder a lot how (much) "your" solution differs from a retail sound card (going by the usual metrics, THD+N, SNR, frequency response) and what more is/would be required except for shielding and maybe a better/other/no codec (since I read somewhere that you said the ported codecs are not 100%, need to re-read that though, not sure anymore).

I will write another post here when I had the time to compare "my solution" here to the UrlStreamBuffered concurrent and non-concurrent.
Cheers!