#include "configuration.h" #include "AudioModule.h" #include "MeshService.h" #include "NodeDB.h" #include "RTC.h" #include "Router.h" #include "FSCommon.h" #include /* AudioModule A interface to send raw codec2 audio data over the mesh network. Based on the example code from the ESP32_codec2 project. https://github.com/deulis/ESP32_Codec2 Codec 2 is a low-bitrate speech audio codec (speech coding) that is patent free and open source develop by David Grant Rowe. http://www.rowetel.com/ and https://github.com/drowe67/codec2 Basic Usage: 1) Enable the module by setting audio.codec2_enabled to 1. 2) Set the pins (audio.mic_pin / audio.amp_pin) for your preferred microphone and amplifier GPIO pins. On tbeam, recommend to use: audio.mic_chan 6 (GPIO 34) audio.amp_pin 14 audio.ptt_pin 39 3) Set audio.timeout to the amount of time to wait before we consider your voice stream as "done". 4) Set audio.bitrate to the desired codec2 rate (CODEC2_3200, CODEC2_2400, CODEC2_1600, CODEC2_1400, CODEC2_1300, CODEC2_1200, CODEC2_700, CODEC2_700B) KNOWN PROBLEMS * Until the module is initilized by the startup sequence, the amp_pin pin is in a floating state. This may produce a bit of "noise". * Will not work on NRF and the Linux device targets. */ #define AMIC 6 #define AAMP 14 #define PTT_PIN 39 #define AUDIO_MODULE_RX_BUFFER 128 #define AUDIO_MODULE_DATA_MAX Constants_DATA_PAYLOAD_LEN #define AUDIO_MODULE_MODE 7 // 700B #define AUDIO_MODULE_ACK 1 #if defined(ARCH_ESP32) && defined(USE_SX1280) AudioModule *audioModule; ButterworthFilter hp_filter(240, 8000, ButterworthFilter::ButterworthFilter::Highpass, 1); //int16_t 1KHz sine test tone int16_t Sine1KHz[8] = { -21210 , -30000, -21210, 0 , 21210 , 30000 , 21210, 0 }; int Sine1KHz_index = 0; uint8_t rx_raw_audio_value = 127; AudioModule::AudioModule() : SinglePortModule("AudioModule", PortNum_AUDIO_APP), concurrency::OSThread("AudioModule") { audio_fifo.init(); } void AudioModule::run_codec2() { if (state == State::tx) { for (int i = 0; i < ADC_BUFFER_SIZE; i++) speech[i] = (int16_t)hp_filter.Update((float)speech[i]); codec2_encode(codec2_state, tx_encode_frame + tx_encode_frame_index, speech); //increment the pointer where the encoded frame must be saved tx_encode_frame_index += 8; //If it is the 5th time then we have a ready trasnmission frame if (tx_encode_frame_index == ENCODE_FRAME_SIZE) { tx_encode_frame_index = 0; //Transmit it sendPayload(); } } if (state == State::rx) //Receiving { //Make a cycle to get each codec2 frame from the received frame for (int i = 0; i < ENCODE_FRAME_SIZE; i += 8) { //Decode the codec2 frame codec2_decode(codec2_state, output_buffer, rx_encode_frame + i); // Add to the audio buffer the 320 samples resulting of the decode of the codec2 frame. for (int g = 0; g < ADC_BUFFER_SIZE; g++) audio_fifo.put(output_buffer[g]); } } state = State::standby; } void AudioModule::handleInterrupt() { audioModule->onTimer(); } void AudioModule::onTimer() { if (state == State::tx) { adc_buffer[adc_buffer_index++] = (16 * adc1_get_raw(mic_chan)) - 32768; //If you want to test with a 1KHz tone, comment the line above and descomment the three lines below // adc_buffer[adc_buffer_index++] = Sine1KHz[Sine1KHz_index++]; // if (Sine1KHz_index >= 8) // Sine1KHz_index = 0; if (adc_buffer_index == ADC_BUFFER_SIZE) { adc_buffer_index = 0; memcpy((void*)speech, (void*)adc_buffer, 2 * ADC_BUFFER_SIZE); audioModule->setIntervalFromNow(0); // process buffer immediately } } else if (state == State::rx) { int16_t v; //Get a value from audio_fifo and convert it to 0 - 255 to play it in the ADC //If none value is available the DAC will play the last one that was read, that's //why the rx_raw_audio_value variable is a global one. if (audio_fifo.get(&v)) rx_raw_audio_value = (uint8_t)((v + 32768) / 256); //Play dacWrite(moduleConfig.audio.amp_pin ? moduleConfig.audio.amp_pin : AAMP, rx_raw_audio_value); } } int32_t AudioModule::runOnce() { if (moduleConfig.audio.codec2_enabled) { if (firstTime) { DEBUG_MSG("Initializing ADC on Channel %u\n", moduleConfig.audio.mic_chan ? moduleConfig.audio.mic_chan : AMIC); mic_chan = moduleConfig.audio.mic_chan ? (adc1_channel_t)(int)moduleConfig.audio.mic_chan : (adc1_channel_t)AMIC; adc1_config_width(ADC_WIDTH_12Bit); adc1_config_channel_atten(mic_chan, ADC_ATTEN_DB_6); // Start a timer at 8kHz to sample the ADC and play the audio on the DAC. uint32_t cpufreq = getCpuFrequencyMhz(); switch (cpufreq){ case 160: adcTimer = timerBegin(3, 1000, true); // 160 MHz / 1000 = 160KHz break; case 240: adcTimer = timerBegin(3, 1500, true); // 240 MHz / 1500 = 160KHz break; case 320: adcTimer = timerBegin(3, 2000, true); // 320 MHz / 2000 = 160KHz break; case 80: default: adcTimer = timerBegin(3, 500, true); // 80 MHz / 500 = 160KHz break; } timerAttachInterrupt(adcTimer, &AudioModule::handleInterrupt, true); timerAlarmWrite(adcTimer, 20, true); // Interrupts when counter == 20, 8.000 times a second timerAlarmEnable(adcTimer); DEBUG_MSG("Initializing DAC on Pin %u\n", moduleConfig.audio.amp_pin ? moduleConfig.audio.amp_pin : AAMP); DEBUG_MSG("Initializing PTT on Pin %u\n", moduleConfig.audio.ptt_pin ? moduleConfig.audio.ptt_pin : PTT_PIN); // Configure PTT input pinMode(moduleConfig.audio.ptt_pin ? moduleConfig.audio.ptt_pin : PTT_PIN, INPUT_PULLUP); state = State::rx; DEBUG_MSG("Setting up codec2 in mode %u\n", moduleConfig.audio.bitrate ? moduleConfig.audio.bitrate : AUDIO_MODULE_MODE); codec2_state = codec2_create(moduleConfig.audio.bitrate ? moduleConfig.audio.bitrate : AUDIO_MODULE_MODE); codec2_set_lpc_post_filter(codec2_state, 1, 0, 0.8, 0.2); firstTime = 0; } else { // Check if we have a PTT press if (digitalRead(moduleConfig.audio.ptt_pin ? moduleConfig.audio.ptt_pin : PTT_PIN) == LOW) { // PTT pressed, recording state = State::tx; } if (state != State::standby) { run_codec2(); } } return 100; } else { DEBUG_MSG("Audio Module Disabled\n"); return INT32_MAX; } } MeshPacket *AudioModule::allocReply() { auto reply = allocDataPacket(); // Allocate a packet for sending return reply; } void AudioModule::sendPayload(NodeNum dest, bool wantReplies) { MeshPacket *p = allocReply(); p->to = dest; p->decoded.want_response = wantReplies; p->want_ack = AUDIO_MODULE_ACK; p->decoded.payload.size = ENCODE_FRAME_SIZE; memcpy(p->decoded.payload.bytes, tx_encode_frame, p->decoded.payload.size); service.sendToMesh(p); } ProcessMessage AudioModule::handleReceived(const MeshPacket &mp) { if (moduleConfig.audio.codec2_enabled) { auto &p = mp.decoded; if (getFrom(&mp) != nodeDB.getNodeNum()) { if (p.payload.size == ENCODE_FRAME_SIZE) { memcpy(rx_encode_frame, p.payload.bytes, p.payload.size); state = State::rx; audioModule->setIntervalFromNow(0); run_codec2(); } else { DEBUG_MSG("Invalid payload size %u != %u\n", p.payload.size, ENCODE_FRAME_SIZE); } } } return ProcessMessage::CONTINUE; } #endif