AR receiver: automatic packet droping

* rework of PR #231
* allow droping packets when
   - first connection
   - silence
   - re-connect after timeout
   - main loop is too slow
This commit is contained in:
Frank
2026-01-11 02:08:41 +01:00
parent 428becc301
commit 201bba0759

View File

@@ -1189,6 +1189,7 @@ class AudioReactive : public Usermod {
};
#define UDPSOUND_MAX_PACKET 96 // max packet size for audiosync, with a bit of "headroom"
#define UDP_AVG_SEND_RATE 23 // 23ms = time for reading one new batch of samples @ 22kHz
// set your config variables to their boot default value (this can also be done in readFromConfig() or a constructor if you prefer)
#if defined(SR_ENABLE_DEFAULT) || defined(UM_AUDIOREACTIVE_ENABLE)
@@ -1848,15 +1849,16 @@ class AudioReactive : public Usermod {
agcSensitivity = 128.0f; // substitute - V1 format does not include this value
}
bool receiveAudioData() {
bool receiveAudioData( unsigned maxSamples) { // maxSamples = 255 means "purge complete input queue"
if (!udpSyncConnected) return false;
bool haveFreshData = false;
size_t packetSize = 0;
static uint8_t fftUdpBuffer[UDPSOUND_MAX_PACKET + 1] = {0};
size_t lastValidPacketSize = 0;
// Loop to read all available packets
while (true) {
// Loop to read available packets
unsigned packetsReceived = 0;
do {
#if __cpp_exceptions
try {
packetSize = fftUdp.parsePacket();
@@ -1867,7 +1869,7 @@ class AudioReactive : public Usermod {
#endif
DEBUG_PRINTLN(F("receiveAudioData: parsePacket out of memory exception caught!"));
USER_FLUSH();
continue; // Skip to next iteration
//continue; // don't skip to next iteration -> we are OOM
}
#else
packetSize = fftUdp.parsePacket();
@@ -1876,31 +1878,35 @@ class AudioReactive : public Usermod {
#ifdef ARDUINO_ARCH_ESP32
if ((packetSize > 0) && ((packetSize < 5) || (packetSize > UDPSOUND_MAX_PACKET))) {
fftUdp.flush();
continue; // Skip invalid packets
continue; // Skip invalid packets -> next iteration
}
#endif
if (packetSize == 0) break; // No more packets available
if (packetSize == 0) break; // No more packets available --> exit loop
if ((packetSize > 5) && (packetSize <= UDPSOUND_MAX_PACKET)) {
fftUdp.read(fftUdpBuffer, packetSize);
lastValidPacketSize = packetSize;
}
}
// Process only the last valid packet
if (lastValidPacketSize > 0) {
if (lastValidPacketSize == sizeof(audioSyncPacket) && (isValidUdpSyncVersion((const char *)fftUdpBuffer))) {
receivedFormat = 2;
haveFreshData = decodeAudioData(lastValidPacketSize, fftUdpBuffer);
} else if (lastValidPacketSize == sizeof(audioSyncPacket_v1) && (isValidUdpSyncVersion_v1((const char *)fftUdpBuffer))) {
decodeAudioData_v1(lastValidPacketSize, fftUdpBuffer);
receivedFormat = 1;
haveFreshData = true;
} else {
receivedFormat = 0; // unknown format
// Process each received packet: last value will persist, intermediate ones needed to update sequence counters
if (lastValidPacketSize > 0) {
if (lastValidPacketSize == sizeof(audioSyncPacket) && (isValidUdpSyncVersion((const char *)fftUdpBuffer))) {
receivedFormat = 2;
haveFreshData |= decodeAudioData(lastValidPacketSize, fftUdpBuffer);
} else if (lastValidPacketSize == sizeof(audioSyncPacket_v1) && (isValidUdpSyncVersion_v1((const char *)fftUdpBuffer))) {
decodeAudioData_v1(lastValidPacketSize, fftUdpBuffer);
receivedFormat = 1;
haveFreshData = true;
} else {
receivedFormat = 0; // unknown format
}
}
}
packetsReceived++;
} while ((packetSize > 0) && ((packetsReceived < maxSamples) || (maxSamples == 255))); // repeat until we have read enough packets, or no more packets available
if ((packetsReceived > 1) && haveFreshData) {USER_PRINTF("AR UDP: dropped %d [%ums\t%d maxDrop].\n", packetsReceived-1, millis() - last_UDPTime, maxSamples-1);} // for debugging
return haveFreshData;
}
@@ -2342,7 +2348,16 @@ class AudioReactive : public Usermod {
bool have_new_sample = false;
if (millis() - lastTime > delayMs) {
// DEBUG_PRINTF(F("AR reading at %d compared to %d max\n"), millis() - lastTime, delayMs); // TroyHacks
have_new_sample = receiveAudioData();
unsigned timeElapsed = (millis() - last_UDPTime);
unsigned maxReadSamples = timeElapsed / UDP_AVG_SEND_RATE; // estimate how many packets can arrived since last receive
maxReadSamples = max(1U, min(maxReadSamples, 20U)); // constrain to [1...20] = max 200ms
// check if we should purge the the receiving queue
if (timeElapsed >= AUDIOSYNC_IDLE_MS) maxReadSamples = 255; // too long since last run
if (receivedFormat == 0) maxReadSamples = 255; // new connection
if (fabsf(volumeSmth) < 0.25f) maxReadSamples = 255; // silence detected
// try to get fresh data
have_new_sample = receiveAudioData(maxReadSamples);
if (have_new_sample) {
last_UDPTime = millis();
useNetworkAudio = true; // UDP input arrived - use it