firmware/src/modules/esp32/AudioModule.cpp

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#include "configuration.h"
#include "AudioModule.h"
#include "MeshService.h"
#include "NodeDB.h"
#include "RTC.h"
#include "Router.h"
#include "FSCommon.h"
#include <assert.h>
/*
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