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WIP: audio module still does not work, but enabled for all regions where audio is permitted.

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# Conflicts:
#	variants/tlora_v2_1_18/platformio.ini
This commit is contained in:
Thomas Göttgens 2022-11-29 11:22:18 +01:00
parent efc3f4c0ee
commit f5120a29ec
5 changed files with 167 additions and 121 deletions
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3
arch/esp32/esp32.ini
View File

@ -33,7 +33,8 @@ lib_deps =
${environmental_base.lib_deps}
https://github.com/meshtastic/esp32_https_server.git#23665b3adc080a311dcbb586ed5941b5f94d6ea2
h2zero/NimBLE-Arduino@^1.4.0
https://github.com/lewisxhe/XPowersLib.git#84b7373faea3118b6c37954d52f98b8a337148d6
https://github.com/lewisxhe/XPowersLib.git#84b7373faea3118b6c37954d52f98b8a337148d6
caveman99/ESP32 Codec2@^1.0.1
lib_ignore =
segger_rtt

1
arch/esp32/esp32s3.ini
View File

@ -34,6 +34,7 @@ lib_deps =
https://github.com/meshtastic/esp32_https_server.git#23665b3adc080a311dcbb586ed5941b5f94d6ea2
h2zero/NimBLE-Arduino@^1.4.0
https://github.com/lewisxhe/XPowersLib.git#84b7373faea3118b6c37954d52f98b8a337148d6
caveman99/ESP32 Codec2@^1.0.1
lib_ignore =
segger_rtt

4
src/modules/Modules.cpp
View File

@ -19,10 +19,8 @@
#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"
#if !defined(TTGO_T_ECHO)
@ -68,9 +66,7 @@ void setupModules()
#endif
#ifdef ARCH_ESP32
// Only run on an esp32 based device.
#ifdef USE_SX1280
new AudioModule();
#endif
new ExternalNotificationModule();
storeForwardModule = new StoreForwardModule();

221
src/modules/esp32/AudioModule.cpp
View File

@ -6,6 +6,8 @@
#include "Router.h"
#include "FSCommon.h"
#include <soc/sens_reg.h>
#include <soc/sens_struct.h>
#include <assert.h>
/*
@ -30,7 +32,7 @@
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".
radio_state. This may produce a bit of "noise".
* Will not work on NRF and the Linux device targets.
*/
@ -40,12 +42,20 @@
#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
#define AUDIO_MODULE_MODE ModuleConfig_AudioConfig_Audio_Baud_CODEC2_700
#if defined(ARCH_ESP32) && defined(USE_SX1280)
#if defined(ARCH_ESP32)
AudioModule *audioModule;
Codec2Thread *codec2Thread;
FastAudioFIFO audio_fifo;
uint16_t adc_buffer[ADC_BUFFER_SIZE] = {};
uint16_t adc_buffer_index = 0;
portMUX_TYPE timerMux = portMUX_INITIALIZER_UNLOCKED;
int16_t speech[ADC_BUFFER_SIZE] = {};
volatile RadioState radio_state = RadioState::tx;
adc1_channel_t mic_chan = (adc1_channel_t)0;
ButterworthFilter hp_filter(240, 8000, ButterworthFilter::ButterworthFilter::Highpass, 1);
@ -55,55 +65,22 @@ int Sine1KHz_index = 0;
uint8_t rx_raw_audio_value = 127;
AudioModule::AudioModule() : SinglePortModule("AudioModule", PortNum_AUDIO_APP), concurrency::OSThread("AudioModule") {
audio_fifo.init();
int IRAM_ATTR local_adc1_read(int channel) {
uint16_t adc_value;
SENS.sar_meas_start1.sar1_en_pad = (1 << channel); // only one channel is selected
while (SENS.sar_slave_addr1.meas_status != 0);
SENS.sar_meas_start1.meas1_start_sar = 0;
SENS.sar_meas_start1.meas1_start_sar = 1;
while (SENS.sar_meas_start1.meas1_done_sar == 0);
adc_value = SENS.sar_meas_start1.meas1_data_sar;
return adc_value;
}
void AudioModule::run_codec2()
IRAM_ATTR void am_onTimer()
{
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;
portENTER_CRITICAL_ISR(&timerMux); //Enter crital code without interruptions
if (radio_state == RadioState::tx) {
adc_buffer[adc_buffer_index++] = (16 * local_adc1_read(mic_chan)) - 32768;
//If you want to test with a 1KHz tone, comment the line above and descomment the three lines below
@ -113,39 +90,96 @@ void AudioModule::onTimer()
if (adc_buffer_index == ADC_BUFFER_SIZE) {
adc_buffer_index = 0;
DEBUG_MSG("--- memcpy\n");
memcpy((void*)speech, (void*)adc_buffer, 2 * ADC_BUFFER_SIZE);
audioModule->setIntervalFromNow(0); // process buffer immediately
// Notify codec2 task that the buffer is ready.
BaseType_t xHigherPriorityTaskWoken = pdFALSE;
DEBUG_MSG("--- notifyFromISR\n");
codec2Thread->notifyFromISR(&xHigherPriorityTaskWoken, RadioState::tx, true);
if (xHigherPriorityTaskWoken)
portYIELD_FROM_ISR();
}
} else if (state == State::rx) {
} else if (radio_state == RadioState::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);
}
portEXIT_CRITICAL_ISR(&timerMux); // exit critical code
}
Codec2Thread::Codec2Thread() : concurrency::NotifiedWorkerThread("Codec2Thread") {
if ((moduleConfig.audio.codec2_enabled) && (myRegion->audioPermitted)) {
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);
} else {
DEBUG_MSG("--- Codec2 disabled\n");
}
}
AudioModule::AudioModule() : SinglePortModule("AudioModule", PortNum_AUDIO_APP), concurrency::OSThread("AudioModule") {
audio_fifo.init();
new Codec2Thread();
}
void Codec2Thread::onNotify(uint32_t notification)
{
switch (notification) {
case RadioState::tx:
for (int i = 0; i < ADC_BUFFER_SIZE; i++)
speech[i] = (int16_t)hp_filter.Update((float)speech[i]);
codec2_encode(codec2_state, audioModule->tx_encode_frame + audioModule->tx_encode_frame_index, speech);
//increment the pointer where the encoded frame must be saved
audioModule->tx_encode_frame_index += 8;
//If it is the 5th time then we have a ready trasnmission frame
if (audioModule->tx_encode_frame_index == ENCODE_FRAME_SIZE)
{
audioModule->tx_encode_frame_index = 0;
//Transmit it
audioModule->sendPayload();
}
break;
case RadioState::rx:
//Make a cycle to get each codec2 frame from the received frame
for (int i = 0; i < ENCODE_FRAME_SIZE; i += ENCODE_CODEC2_SIZE)
{
//Decode the codec2 frame
codec2_decode(codec2_state, output_buffer, audioModule->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]);
}
break;
default:
assert(0); // We expected to receive a valid notification from the ISR
break;
}
}
int32_t AudioModule::runOnce()
{
if (moduleConfig.audio.codec2_enabled) {
if ((moduleConfig.audio.codec2_enabled) && (myRegion->audioPermitted)) {
if (firstTime) {
DEBUG_MSG("Initializing ADC on Channel %u\n", moduleConfig.audio.mic_chan ? moduleConfig.audio.mic_chan : AMIC);
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);
adc1_config_channel_atten(mic_chan, ADC_ATTEN_DB_6);
adc1_get_raw(mic_chan);
radio_state = RadioState::rx;
// Start a timer at 8kHz to sample the ADC and play the audio on the DAC.
uint32_t cpufreq = getCpuFrequencyMhz();
switch (cpufreq){
switch (cpufreq){
case 160:
adcTimer = timerBegin(3, 1000, true); // 160 MHz / 1000 = 160KHz
break;
@ -160,48 +194,44 @@ int32_t AudioModule::runOnce()
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("--- Timer CPU Frequency: %u MHz\n", cpufreq);
timerAttachInterrupt(adcTimer, &am_onTimer, false);
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);
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);
pinMode(moduleConfig.audio.ptt_pin ? moduleConfig.audio.ptt_pin : PTT_PIN, INPUT);
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;
firstTime = false;
} 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;
// Check if PTT is pressed. TODO hook that into Onebutton/Interrupt drive.
if (digitalRead(moduleConfig.audio.ptt_pin ? moduleConfig.audio.ptt_pin : PTT_PIN) == HIGH) {
if (radio_state == RadioState::rx) {
DEBUG_MSG("--- PTT pressed, switching to TX\n");
radio_state = RadioState::tx;
}
} else {
if (radio_state == RadioState::tx) {
DEBUG_MSG("--- PTT released, switching to RX\n");
radio_state = RadioState::rx;
}
}
if (state != State::standby) {
run_codec2();
}
}
}
return 100;
} else {
DEBUG_MSG("Audio Module Disabled\n");
DEBUG_MSG("--- Audio Module Disabled\n");
return INT32_MAX;
}
}
MeshPacket *AudioModule::allocReply()
{
auto reply = allocDataPacket(); // Allocate a packet for sending
return reply;
}
@ -211,7 +241,8 @@ void AudioModule::sendPayload(NodeNum dest, bool wantReplies)
p->to = dest;
p->decoded.want_response = wantReplies;
p->want_ack = AUDIO_MODULE_ACK;
p->want_ack = false; // Audio is shoot&forget. TODO: Is this really suppressing retransmissions?
p->priority = MeshPacket_Priority_MAX; // Audio is important, because realtime
p->decoded.payload.size = ENCODE_FRAME_SIZE;
memcpy(p->decoded.payload.bytes, tx_encode_frame, p->decoded.payload.size);
@ -221,16 +252,18 @@ void AudioModule::sendPayload(NodeNum dest, bool wantReplies)
ProcessMessage AudioModule::handleReceived(const MeshPacket &mp)
{
if (moduleConfig.audio.codec2_enabled) {
if ((moduleConfig.audio.codec2_enabled) && (myRegion->audioPermitted)) {
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();
radio_state = RadioState::rx;
BaseType_t xHigherPriorityTaskWoken = pdFALSE;
codec2Thread->notifyFromISR(&xHigherPriorityTaskWoken, RadioState::rx, true);
if (xHigherPriorityTaskWoken)
portYIELD_FROM_ISR();
} else {
DEBUG_MSG("Invalid payload size %u != %u\n", p.payload.size, ENCODE_FRAME_SIZE);
DEBUG_MSG("--- Invalid payload size %u != %u\n", p.payload.size, ENCODE_FRAME_SIZE);
}
}
}

59
src/modules/esp32/AudioModule.h
View File

@ -7,38 +7,45 @@
#include <Arduino.h>
#include <driver/adc.h>
#include <functional>
#if defined(ARCH_ESP32) && defined(USE_SX1280)
#if defined(ARCH_ESP32)
#include <codec2.h>
#include <ButterworthFilter.h>
#include <FastAudioFIFO.h>
#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
#define ENCODE_CODEC2_SIZE 8
#define ENCODE_FRAME_SIZE (ENCODE_CODEC2_SIZE * 5) // 5 codec2 frames of 8 bytes each
class Codec2Thread : public concurrency::NotifiedWorkerThread
{
#if defined(ARCH_ESP32)
struct CODEC2* codec2_state = NULL;
int16_t output_buffer[ADC_BUFFER_SIZE] = {};
public:
Codec2Thread();
protected:
virtual void onNotify(uint32_t notification) override;
#endif
};
class AudioModule : public SinglePortModule, private concurrency::OSThread
{
#if defined(ARCH_ESP32) && defined(USE_SX1280)
bool firstTime = 1;
#if defined(ARCH_ESP32)
bool firstTime = true;
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 = NULL;
enum State
{
standby, rx, tx
};
volatile State state = State::tx;
public:
unsigned char rx_encode_frame[ENCODE_FRAME_SIZE] = {};
unsigned char tx_encode_frame[ENCODE_FRAME_SIZE] = {};
int tx_encode_frame_index = 0;
AudioModule();
/**
@ -49,11 +56,7 @@ class AudioModule : public SinglePortModule, private concurrency::OSThread
protected:
virtual int32_t runOnce() override;
static void handleInterrupt();
void onTimer();
void run_codec2();
// void run_codec2();
virtual MeshPacket *allocReply() override;
@ -65,4 +68,16 @@ class AudioModule : public SinglePortModule, private concurrency::OSThread
};
extern AudioModule *audioModule;
extern Codec2Thread *codec2Thread;
extern FastAudioFIFO audio_fifo;
extern uint16_t adc_buffer[ADC_BUFFER_SIZE];
extern uint16_t adc_buffer_index;
extern portMUX_TYPE timerMux;
extern int16_t speech[ADC_BUFFER_SIZE];
enum RadioState { standby, rx, tx };
extern volatile RadioState radio_state;
extern adc1_channel_t mic_chan;
IRAM_ATTR void am_onTimer();