codec2 Audio Support for SX1280 (untested)

This commit is contained in:
Thomas Göttgens 2022-11-02 18:23:41 +01:00
parent ef18b173cd
commit 09ddde177c
4 changed files with 325 additions and 5 deletions

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@ -19,6 +19,9 @@
#ifdef ARCH_ESP32
#include "modules/esp32/RangeTestModule.h"
#include "modules/esp32/StoreForwardModule.h"
#ifdef USE_SX1280
#include "modules/esp32/AudioModule.h"
#endif
#endif
#if defined(ARCH_ESP32) || defined(ARCH_NRF52)
#include "modules/ExternalNotificationModule.h"
@ -65,17 +68,16 @@ void setupModules()
#endif
#ifdef ARCH_ESP32
// Only run on an esp32 based device.
/*
Maintained by MC Hamster (Jm Casler) jm@casler.org
*/
#ifdef USE_SX1280
new AudioModule();
#endif
new ExternalNotificationModule();
storeForwardModule = new StoreForwardModule();
new RangeTestModule();
#elif defined(ARCH_NRF52)
new ExternalNotificationModule();
new ExternalNotificationModule();
#endif
// NOTE! This module must be added LAST because it likes to check for replies from other modules and avoid sending extra acks

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@ -0,0 +1,249 @@
#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 7 (GPIO 35)
audio.amp_pin 25 (GPIO 25)
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 7
#define AAMP 25
#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
AudioModule *audioModule;
#if defined(ARCH_ESP32) && defined(USE_SX1280)
ButterworthFilter hp_filter(240, 8000, ButterworthFilter::ButterworthFilter::Highpass, 1);
#endif
//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") {}
void AudioModule::run_codec2()
{
#if defined(ARCH_ESP32) && defined(USE_SX1280)
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;
#endif
}
void AudioModule::handleInterrupt()
{
audioModule->onTimer();
}
void AudioModule::onTimer()
{
#if defined(ARCH_ESP32) && defined(USE_SX1280)
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);
}
#endif
}
int32_t AudioModule::runOnce()
{
#if defined(ARCH_ESP32) && defined(USE_SX1280)
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;
}
#else
return INT32_MAX;
#endif
}
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 defined(ARCH_ESP32) && defined(USE_SX1280)
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);
}
}
}
#endif
return ProcessMessage::CONTINUE;
}

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@ -0,0 +1,68 @@
#pragma once
#include "SinglePortModule.h"
#include "concurrency/OSThread.h"
#include "configuration.h"
#include "NodeDB.h"
#include <Arduino.h>
#include <driver/adc.h>
#include <functional>
#if defined(ARCH_ESP32) && defined(USE_SX1280)
#include <codec2.h>
#include <ButterworthFilter.h>
#include <FastAudioFIFO.h>
#else
typedef int FastAudioFIFO;
#endif
#define ADC_BUFFER_SIZE 320 // 40ms of voice in 8KHz sampling frequency
#define ENCODE_FRAME_SIZE 40 // 5 codec2 frames of 8 bytes each
class AudioModule : public SinglePortModule, private concurrency::OSThread
{
bool firstTime = 1;
hw_timer_t* adcTimer = NULL;
uint16_t adc_buffer[ADC_BUFFER_SIZE];
int16_t speech[ADC_BUFFER_SIZE];
int16_t output_buffer[ADC_BUFFER_SIZE];
unsigned char rx_encode_frame[ENCODE_FRAME_SIZE];
unsigned char tx_encode_frame[ENCODE_FRAME_SIZE];
int tx_encode_frame_index = 0;
FastAudioFIFO audio_fifo;
uint16_t adc_buffer_index = 0;
adc1_channel_t mic_chan = (adc1_channel_t)0;
struct CODEC2* codec2_state;
enum State
{
standby, rx, tx
};
volatile State state = State::tx;
public:
AudioModule();
/**
* Send our payload into the mesh
*/
void sendPayload(NodeNum dest = NODENUM_BROADCAST, bool wantReplies = false);
protected:
virtual int32_t runOnce() override;
static void handleInterrupt();
void onTimer();
void run_codec2();
virtual MeshPacket *allocReply() override;
/** Called to handle a particular incoming message
* @return ProcessMessage::STOP if you've guaranteed you've handled this message and no other handlers should be considered for it
*/
virtual ProcessMessage handleReceived(const MeshPacket &mp) override;
};
extern AudioModule *audioModule;

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@ -3,5 +3,6 @@ extends = esp32_base
board = ttgo-lora32-v21
lib_deps =
${esp32_base.lib_deps}
caveman99/ESP32 Codec2@^1.0.1
build_flags =
${esp32_base.build_flags} -D TLORA_V2_1_18 -I variants/tlora_v2_1_18