When I first tried the arduino code with my mpu6050, the serial monitor showed the readings. However, when I tried to connect it with unity, the readings cannot be showed or used in unity. I was stuck at this problem. I was trying to make the cube that i created in unity to move with the mouseposition then replaced by the mpu 6050.
Arduino code
#include <Wire.h>
#include <I2Cdev.h>
#include <MPU6050.h>
MPU6050 mpu;
int16_t ax, ay, az, gx, gy, gz;
int vx, vy;
void setup() {
Serial.begin(115200);
Wire.begin();
mpu.initialize();
//if (!mpu.testConnection()) { while (1); }
}
void loop() {
mpu.getMotion6(&ax, &ay, &az, &gx, &gy, &gz);
vx = ((gx+300)/150)-1; // "+300" because the x axis of gyroscope give
values about -350 while it's not moving. Change this value if you get
something different using the TEST code, chacking if there are values far
from zero.
vy = -(gz-100)/150; // same here about "-100"
String output = String(gx) + ";" + String(gz) + ";" + String(vx) + ";" +
String(vy);
Serial.println(output);
Serial.flush();
delay(10);
}
Unity code
public class Cube : MonoBehaviour {
SerialPort stream = new SerialPort("COM4", 115200);
public Camera cam;
public float mousex;
public float mousey;
public GameObject target; // is the gameobject to
public float acc_normalizer_factor = 0.00025f;
public float gyro_normalizer_factor = 1.0f / 32768.0f; // 32768 is max
value captured during test on imu
//float curr_angle_x = 0;
//float curr_angle_y = 0;
//float curr_angle_z = 0;
float curr_offset_x = 0;
float curr_offset_y = -2;
void Start()
{
stream.Open();
cam = Camera.main;
}
void Update()
{
string dataString = "null received";
dataString = stream.ReadLine();
char splitChar = ';';
string[] dataRaw = dataString.Split(splitChar);
float vx = int.Parse(dataRaw[2]);
float vy = int.Parse(dataRaw[3]);
curr_offset_x += (vx * acc_normalizer_factor);
curr_offset_y += (vy * acc_normalizer_factor);
//get mouse possition
mousex = (curr_offset_x);
mousey = (curr_offset_y);
//adapt it to screen
// 5 in z to be in the front of the camera, but up to you, just
avoid 0
Vector3 mouseposition = cam.ScreenToWorldPoint(new Vector3(mousex,
mousey, 5));
//make your gameobject fallow your input
target.transform.position = mouseposition;
}
First of all, check your device. Is the data being sent by your device and received by machine the mpu is connected to?
Also you can try to Debug.Log(dataString) in Unity right after dataString = stream.ReadLine(); in your code (it makes sense only if data from mpu is receiving by machine your device is connected to).
I need to use Bitmap class from System.Drawing.Bitmap, this is a function that works fine on windows Platform. But after I tried to run on Xamarin Forms, and installed nuget package System.Drawing, the program compiles correctly without errors.
But when running program I receive an error. Somehow seems to point to System.Drawing from windows, not the System.Drawing from the nuget package.
What I need to do is, get Photo from Camera and print it.
Below is the code to print. Problem is with "Bitmap" converter.
Tried several nuget packages, none worked:
System.Drawing.Common
Fast-Bitmap
Bitmap.Net
public byte[] PrintImage(byte[] PHOTO)
{
Bitmap bmp;
using (var ms = new MemoryStream(PHOTO))
{
bmp = new Bitmap(ms);
}
BitmapData data = GetBitmapData(bmp);
BitArray dots = data.Dots;
byte[] width = BitConverter.GetBytes(data.Width);
int offset = 0;
MemoryStream stream = new MemoryStream();
BinaryWriter bw = new BinaryWriter(stream);
// center command
bw.Write(27);
bw.Write('a');
bw.Write(1);
// print image
bw.Write((char)0x1B);
bw.Write('#');
bw.Write((char)0x1B);
bw.Write('3');
bw.Write((byte)24);
while (offset < data.Height)
{
bw.Write((char)0x1B);
bw.Write('*'); // bit-image mode
bw.Write((byte)33); // 24-dot double-density
bw.Write(width[0]); // width low byte
bw.Write(width[1]); // width high byte
for (int x = 0; x < data.Width; ++x)
{
for (int k = 0; k < 3; ++k)
{
byte slice = 0;
for (int b = 0; b < 8; ++b)
{
int y = (((offset / 8) + k) * 8) + b;
// Calculate the location of the pixel.
// It'll be at (y * width) + x.
int i = (y * data.Width) + x;
// If the image is shorter than 24 dots.
bool v = false;
if (i < dots.Length)
{
v = dots[i];
}
slice |= (byte)((v ? 1 : 0) << (7 - b));
}
bw.Write(slice);
}
}
offset += 24;
bw.Write((char)0x0A);
}
// Restore the line spacing to the default of 30 dots.
bw.Write((char)0x1B);
bw.Write('3');
bw.Write((byte)30);
bw.Flush();
byte[] bytes = stream.ToArray();
return bytes; // logo + Encoding.Default.GetString(bytes);
}
public BitmapData GetBitmapData(Bitmap bmp) // (string bmpFileName)
{
//using (var bitmap = (Bitmap)Bitmap.FromFile(bmpFileName))
using (var bitmap = bmp)
{
var threshold = 127;
var index = 0;
double multiplier = 570; // this depends on your printer
double scale = (double)(multiplier / (double)bitmap.Width);
int xheight = (int)(bitmap.Height * scale);
int xwidth = (int)(bitmap.Width * scale);
var dimensions = xwidth * xheight;
var dots = new BitArray(dimensions);
for (var y = 0; y < xheight; y++)
{
for (var x = 0; x < xwidth; x++)
{
var _x = (int)(x / scale);
var _y = (int)(y / scale);
var color = bitmap.GetPixel(_x, _y);
var luminance = (int)(color.R * 0.3 + color.G * 0.59 + color.B * 0.11);
dots[index] = (luminance < threshold);
index++;
}
}
return new BitmapData()
{
Dots = dots,
Height = (int)(bitmap.Height * scale),
Width = (int)(bitmap.Width * scale)
};
}
}
public class BitmapData
{
public BitArray Dots
{
get;
set;
}
public int Height
{
get;
set;
}
public int Width
{
get;
set;
}
}
Error occurs when function is called as:
byte[] _buffer = PrintImage(FOTO);
The error:
"Could not resolve type with token 01000119 from typeref (expected class 'System.Drawing.Bitmap' in assembly 'System.Drawing.Common, Version=4.0.1.0, Culture=neutral, PublicKeyToken=cc7b13ffcd2ddd51')"
I am making a simple voice visualization program. My goals are:
Playback microphone input
Visualize voice spectrum and gain in real time
Here is my code:
using System.Collections;
using System.Collections.Generic;
using UnityEngine;
public class VisualizeVoice : MonoBehaviour
{
private const int NUM_SPECTRUM_SAMPLES = 256;
private const int NUM_SPECTRUM_BARS = 32;
private const int NUM_PCM_SAMPLES = 16000;
private const float BAR_DROP_SPEED = 1e-3f;
private const int NUM_SAMPLES_TO_AVERAGE = 8;
private string _deviceName;
private float[] _spectrumData = new float[NUM_SPECTRUM_SAMPLES];
private float[] _fPCMData = new float[NUM_PCM_SAMPLES];
private float _gain = 0;
private AudioClip _audio; // Audio from microphone
private AudioSource _playback; // To play the audio from microphone
// For visualization
private GameObject[] _spectrumBars = new GameObject[NUM_SPECTRUM_BARS];
private GameObject _gainBar;
// Start is called before the first frame update
void Start()
{
if (Microphone.devices.Length == 0) {
Debug.LogError("No Microphone");
return;
}
_deviceName = Microphone.devices[0];
Debug.Log("Current microphone is " + _deviceName);
if ((_playback = this.GetComponent<AudioSource>()) == null) {
_playback = this.gameObject.AddComponent<AudioSource>();
}
_playback.loop = true;
_playback.bypassEffects = true;
_playback.bypassListenerEffects = true;
_playback.bypassReverbZones = true;
_playback.priority = 0;
_playback.pitch = 1;
_playback.clip = _audio = Microphone.Start(_deviceName, true, 1, AudioSettings.outputSampleRate);
// Sync microphone and playback, but it always fails
float waitTime = 0;
while (!(Microphone.GetPosition(_deviceName) > 0) && waitTime <= 2)
waitTime += Time.deltaTime;
if (waitTime > 2) {
Debug.LogError("time out waiting for microphone");
}
_playback.Play();
InitVisualization();
}
// Update is called once per frame
void Update()
{
// Get PCM data and calculate gain
var audioPosition = Microphone.GetPosition(_deviceName);
_audio.GetData(_fPCMData, audioPosition);
UpdateGain();
// Get spectrum data
_playback.GetSpectrumData(_spectrumData, 0, FFTWindow.BlackmanHarris);
// Update visualization
UpdateVisualization();
}
private void InitVisualization()
{
// Initialize spectrum bars
for (int ibar = 0; ibar < NUM_SPECTRUM_BARS; ibar++) {
_spectrumBars[ibar] = GameObject.CreatePrimitive(PrimitiveType.Cube);
_spectrumBars[ibar].transform.parent = this.transform;
_spectrumBars[ibar].transform.localPosition = new Vector3(ibar, 0, 0);
_spectrumBars[ibar].transform.localScale = new Vector3(1, 0, 1);
}
// Initialize gain bar
_gainBar = GameObject.CreatePrimitive(PrimitiveType.Cube);
_gainBar.transform.parent = this.transform;
_gainBar.transform.localPosition = new Vector3(-5, 0, 0);
_gainBar.transform.localScale = new Vector3(4, 0, 1);
// Overall dimension
this.transform.localScale = new Vector3(0.2f, 10.0f, 0.2f);
}
private void UpdateVisualization()
{
// Update spectrum bars
int nSamplesPerBar = NUM_SPECTRUM_SAMPLES / NUM_SPECTRUM_BARS;
for (int ibar = 0; ibar < NUM_SPECTRUM_BARS; ibar++) {
// Calculate value of each bar
float value = 0;
for (int isample = 0; isample < nSamplesPerBar; isample++) {
value += _spectrumData[ibar * nSamplesPerBar + isample];
}
value /= nSamplesPerBar;
// Use current value if increasing, or slowly drop previous value if decreasing
float prevValue = _spectrumBars[ibar].transform.localScale.y;
if (value < prevValue)
value = prevValue - BAR_DROP_SPEED;
// Y scale is set to value
_spectrumBars[ibar].transform.localScale = new Vector3(1, value, 1);
}
// Update gain bar
_gainBar.transform.localScale = new Vector3(4, _gain, 1);
}
private void UpdateGain()
{
_gain = 0;
for(int i = 0; i < NUM_SAMPLES_TO_AVERAGE; i++) {
_gain += Mathf.Abs(_fPCMData[NUM_PCM_SAMPLES - i - 1]);
}
_gain /= NUM_SAMPLES_TO_AVERAGE;
}
}
Here are my questions:
I can't use while (!Microphone.GetPosition(_deviceName) > 0)); to avoid latency from microphone to speaker. If I use it, my application just freezes. If I add code to allow time-out, it has time-out every time.
The gain bar seems irrelevant with my voice. I don't know if my calculation is right.
I'm not sure if I need to average over multiple samples calculating gains, and how many samples I need to average over. I need this gain value later to detect silent moments and cut audio data.
To 1.
You can. Unity allows to define Start as a Coroutine
private IEnumerator Start()
{
...
}
On this way you can use a non blocking
while (!Microphone.GetPosition(_deviceName) > 0))
{
yield return null;
}
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I a doing a weird project, and looking to convert some short, simple datagrams to audio - send them over a (physical) radio - then to receive and decode them on another device (think - embedded devices with audio out jack and GSM/GPRS-type radios).
(I have to use a physical, existing external radio).
Does anyone know of a good, simple software modem library good for such a project? I'm not so concerned about data rate, and would prefer simplicity over functionality. Even something akin to a basic 1200 baud modem would be fantastic.
Looking at this more of a learning experience and potential building block, rather than anything horribly practical.
As an exercise I've implemented a simple V.23-like modem using FSK modulation and supporting a data rate of 1200 bits/second (960 bits/second effective because of the start and stop bits).
I'm curious to see if it works with your radio. Noise, signal reflection and imperfect demodulation can all affect the performance of the modem.
Prior to trying to integrate this code into your project, first see if it works with audio recorded from your radio.
Code:
#include <stdio.h>
#include <string.h>
#include <stdlib.h>
#include <limits.h>
#include <math.h>
#ifndef M_PI
#define M_PI 3.14159265358979324
#endif
typedef unsigned char uchar, uint8;
typedef signed char schar, int8;
typedef unsigned short ushort, uint16;
typedef short int16;
typedef unsigned int uint;
typedef unsigned long ulong;
#if UINT_MAX >= 0xFFFFFFFF
typedef int int32;
typedef unsigned int uint32;
#else
typedef long int32;
typedef unsigned long uint32;
#endif
typedef long long int64;
typedef unsigned long long uint64;
typedef struct
{
double x, y;
} tComplex;
tComplex complexAdd(const tComplex* a, const tComplex* b)
{
tComplex c;
c.x = a->x + b->x;
c.y = a->y + b->y;
return c;
}
tComplex complexMul(const tComplex* a, const tComplex* b)
{
tComplex c;
c.x = a->x * b->x - a->y * b->y;
c.y = a->x * b->y + a->y * b->x;
return c;
}
void dft(tComplex out[], const tComplex in[], size_t n, int direction)
{
size_t k, i;
for (k = 0; k < n; k++)
{
tComplex r = { 0, 0 }, e;
for (i = 0; i < n; i++)
{
e.x = cos(-2 * direction * M_PI / n * ((double)k - n / 2) * ((double)i - n / 2));
e.y = sin(-2 * direction * M_PI / n * ((double)k - n / 2) * ((double)i - n / 2));
e = complexMul(&e, &in[i]);
r = complexAdd(&r, &e);
}
out[k] = r;
}
}
#define FILTER_LENGTH 64
typedef struct tTx
{
enum
{
stSendingOnes,
stSendingData
} State;
uint SampleRate;
uint OnesFreq;
uint ZeroesFreq;
uint BitRate;
uint32 SampleCnt;
uint BitSampleCnt;
uint Data;
uint DataLeft;
double Phase;
double PhaseIncrement;
uint (*pTxGetDataCallBack)(struct tTx*, uint8*);
} tTx;
void TxInit(tTx* pTx,
uint SampleRate,
uint (*pTxGetDataCallBack)(tTx*, uint8*))
{
memset(pTx, 0, sizeof(*pTx));
pTx->State = stSendingOnes;
pTx->SampleRate = SampleRate;
pTx->OnesFreq = 1300;
pTx->ZeroesFreq = 2100;
pTx->BitRate = 1200;
pTx->pTxGetDataCallBack = pTxGetDataCallBack;
pTx->SampleCnt = 0;
pTx->BitSampleCnt = pTx->SampleRate;
pTx->Data = 0;
pTx->DataLeft = 0;
pTx->Phase = 0.0;
pTx->PhaseIncrement = 2 * M_PI * pTx->OnesFreq / pTx->SampleRate;
}
int16 TxGetSample(tTx* pTx)
{
int16 sample;
if (pTx->State == stSendingOnes &&
pTx->SampleCnt >= pTx->SampleRate)
{
// Sent 1 second worth of 1's, can now send data
pTx->State = stSendingData;
}
if (pTx->State == stSendingData &&
pTx->BitSampleCnt >= pTx->SampleRate)
{
// Another data bit can now be sent
uint8 d;
pTx->BitSampleCnt -= pTx->SampleRate;
if (!pTx->DataLeft)
{
// Get the next data byte (if any)
if (pTx->pTxGetDataCallBack(pTx, &d) != 0)
{
pTx->Data = d & 0xFF;
pTx->Data |= 1 << 8; // insert the stop bit
pTx->Data <<= 1; // insert the start bit
pTx->DataLeft = 10;
}
else
{
pTx->Data = 0x3FF; // no data, send 10 1's
pTx->DataLeft = 10;
}
}
// Extract the next data bit to send
d = pTx->Data & 1;
pTx->Data >>= 1;
pTx->DataLeft--;
// Choose the appropriate frequency for 0 and 1
if (d)
{
pTx->PhaseIncrement = 2 * M_PI * pTx->OnesFreq / pTx->SampleRate;
}
else
{
pTx->PhaseIncrement = 2 * M_PI * pTx->ZeroesFreq / pTx->SampleRate;
}
}
// Generate the next sample, advance the generator's phase
sample = (int16)(16000 * cos(pTx->Phase));
pTx->Phase += pTx->PhaseIncrement;
if (pTx->Phase >= 2 * M_PI)
{
pTx->Phase -= 2 * M_PI;
}
if (pTx->State == stSendingData)
{
pTx->BitSampleCnt += pTx->BitRate;
}
pTx->SampleCnt++;
return sample;
}
typedef struct tRx
{
enum
{
stCarrierLost,
stCarrierDetected,
stReceivingData
} State;
uint SampleRate;
uint OnesFreq;
uint ZeroesFreq;
uint MidFreq;
uint BitRate;
uint32 SampleCnt;
uint BitSampleCnt;
uint Data;
double Phase;
double PhaseIncrement;
tComplex Filter[FILTER_LENGTH];
double Delay[FILTER_LENGTH];
double LastAngle;
int LastDelta;
int32 Deltas;
int32 CarrierAngle;
int32 CarrierCnt;
double LongAvgPower;
double ShortAvgPower;
void (*pRxGetDataCallBack)(struct tRx*, uint8);
} tRx;
void RxInit(tRx* pRx,
uint SampleRate,
void (*pRxGetDataCallBack)(struct tRx*, uint8))
{
tComplex tmp[FILTER_LENGTH];
uint i;
memset(pRx, 0, sizeof(*pRx));
pRx->State = stCarrierLost;
pRx->SampleRate = SampleRate;
pRx->OnesFreq = 1300;
pRx->ZeroesFreq = 2100;
pRx->MidFreq = (pRx->OnesFreq + pRx->ZeroesFreq) / 2;
pRx->BitRate = 1200;
pRx->pRxGetDataCallBack = pRxGetDataCallBack;
pRx->SampleCnt = 0;
pRx->BitSampleCnt = 0;
pRx->Data = 0x3FF;
pRx->Phase = 0.0;
pRx->PhaseIncrement = 2 * M_PI * pRx->MidFreq / pRx->SampleRate;
pRx->LastAngle = 0.0;
pRx->LastDelta = 0;
pRx->Deltas = 0;
pRx->CarrierAngle = 0;
pRx->CarrierCnt = 0;
pRx->LongAvgPower = 0.0;
pRx->ShortAvgPower = 0.0;
for (i = 0; i < FILTER_LENGTH; i++)
{
pRx->Delay[i] = 0.0;
}
for (i = 0; i < FILTER_LENGTH; i++)
{
if (i == 0) // w < 0 (min w)
{
pRx->Filter[i].x = 0;
pRx->Filter[i].y = 0;
}
else if (i < FILTER_LENGTH / 2) // w < 0
{
pRx->Filter[i].x = 0;
pRx->Filter[i].y = 0;
}
else if (i == FILTER_LENGTH / 2) // w = 0
{
pRx->Filter[i].x = 0;
pRx->Filter[i].y = 0;
}
else if (i > FILTER_LENGTH / 2) // w > 0
{
pRx->Filter[i].x = 0;
pRx->Filter[i].y = -1;
// Extra filter to combat channel noise
if (i - FILTER_LENGTH / 2 < 875UL * FILTER_LENGTH / pRx->SampleRate ||
i - FILTER_LENGTH / 2 > (2350UL * FILTER_LENGTH + pRx->SampleRate - 1) / pRx->SampleRate)
{
pRx->Filter[i].y = 0;
}
}
}
memcpy(tmp, pRx->Filter, sizeof(tmp));
dft(pRx->Filter, tmp, FILTER_LENGTH, -1);
}
#define RX_VERBOSE 0
void RxGetSample(tRx* pRx, int16 Sample)
{
tComplex s = { 0.0, 0.0 }, ss;
double angle;
uint i;
int delta;
double pwr;
// Insert the sample into the delay line
memmove(&pRx->Delay[0], &pRx->Delay[1], sizeof(pRx->Delay) - sizeof(pRx->Delay[0]));
pRx->Delay[FILTER_LENGTH - 1] = Sample;
// Get the next analytic signal sample by applying Hilbert transform/filter
for (i = 0; i < FILTER_LENGTH; i++)
{
s.x += pRx->Delay[i] * pRx->Filter[FILTER_LENGTH - 1 - i].x;
s.y += pRx->Delay[i] * pRx->Filter[FILTER_LENGTH - 1 - i].y;
}
// Frequency shift by MidFreq down
ss.x = cos(-pRx->Phase);
ss.y = sin(-pRx->Phase);
s = complexMul(&s, &ss);
pRx->Phase += pRx->PhaseIncrement;
if (pRx->Phase >= 2 * M_PI)
{
pRx->Phase -= 2 * M_PI;
}
// Calculate signal power
pwr = (s.x * s.x + s.y * s.y) / 32768 / 32768;
pRx->LongAvgPower *= 1 - pRx->BitRate / (pRx->SampleRate * 8.0 * 8);
pRx->LongAvgPower += pwr;
pRx->ShortAvgPower *= 1 - pRx->BitRate / (pRx->SampleRate * 8.0);
pRx->ShortAvgPower += pwr;
#if 0
printf("LongAvgPower:%f ShortAvgPower:%f\n", pRx->LongAvgPower, pRx->ShortAvgPower);
#endif
// Disconnect if the signal power changes abruptly.
if (pRx->State != stCarrierLost &&
pRx->LongAvgPower > pRx->ShortAvgPower * 8 * 8)
{
// N.B. The receiver may have received a few extra (garbage) bytes
// while demodulating the abruptly changed signal.
// Prefixing data with its size or using a more advanced protocol
// may be a good solution to this little problem.
pRx->State = stCarrierLost;
pRx->BitSampleCnt = 0;
pRx->Data = 0x3FF;
pRx->Phase = 0.0;
pRx->LastAngle = 0.0;
pRx->LastDelta = 0;
pRx->Deltas = 0;
pRx->CarrierAngle = 0;
pRx->CarrierCnt = 0;
}
// Get the phase angle from the analytic signal sample
angle = (fpclassify(s.x) == FP_ZERO && fpclassify(s.y) == FP_ZERO) ?
0.0 : 180 / M_PI * atan2(s.y, s.x);
// Calculate the phase angle change and force it to the -PI to +PI range
delta = (int)(360.5 + angle - pRx->LastAngle) % 360;
if (delta > 180) delta -= 360;
if (pRx->State == stCarrierLost)
{
// Accumulate the phase angle change to see if we're receiving 1's
pRx->CarrierAngle += delta;
pRx->CarrierCnt++;
// Check whether or not the phase corresponds to 1's
if (delta < 0)
{
if (pRx->CarrierCnt >= pRx->SampleRate / pRx->OnesFreq * 8)
{
double ph = (double)pRx->CarrierAngle / pRx->CarrierCnt;
#if RX_VERBOSE
printf("ca:%5ld, cc:%4ld, ca/cc:%4ld\n",
(long)pRx->CarrierAngle,
(long)pRx->CarrierCnt,
(long)(pRx->CarrierAngle / pRx->CarrierCnt));
#endif
// Frequency tolerance is +/-16 Hz per the V.23 spec
if (ph < (pRx->OnesFreq - 17.0 - pRx->MidFreq) * 360.0 / pRx->SampleRate ||
ph > (pRx->OnesFreq + 17.0 - pRx->MidFreq) * 360.0 / pRx->SampleRate)
{
goto BadCarrier;
}
}
}
else
{
BadCarrier:
// Phase doesn't correspond to 1's
pRx->CarrierAngle = 0.0;
pRx->CarrierCnt = 0;
}
if (pRx->CarrierCnt >= pRx->SampleRate / 2 + pRx->SampleRate / 4)
{
// 0.75 seconds worth of 1's have been detected, ready to receive data
// Adjust MidFreq to compensate for the DAC and ADC sample rate difference
double f1 = (double)pRx->CarrierAngle / pRx->CarrierCnt / 360 * pRx->SampleRate + pRx->MidFreq;
pRx->MidFreq = (uint)(pRx->MidFreq * f1 / pRx->OnesFreq);
pRx->PhaseIncrement = 2 * M_PI * pRx->MidFreq / pRx->SampleRate;
#if RX_VERBOSE
printf("f1:%u, new MidFreq:%u\n", (uint)f1, pRx->MidFreq);
#endif
pRx->State = stCarrierDetected;
}
}
else
{
// Detect frequency changes (transitions between 0's and 1's)
int freqChange = ((int32)pRx->LastDelta * delta < 0 || pRx->LastDelta && !delta);
int reAddDelta = 0;
#if RX_VERBOSE
printf("%6lu: delta:%4d freqChange:%d BitSampleCnt:%u\n",
(ulong)pRx->SampleCnt,
delta,
freqChange,
pRx->BitSampleCnt);
#endif
// Synchronize with 1<->0 transitions
if (freqChange)
{
if (pRx->BitSampleCnt >= pRx->SampleRate / 2)
{
pRx->BitSampleCnt = pRx->SampleRate;
pRx->Deltas -= delta;
reAddDelta = 1;
}
else
{
pRx->BitSampleCnt = 0;
pRx->Deltas = 0;
}
}
// Accumulate analytic signal phase angle changes
// (positive for 0, negative for 1)
pRx->Deltas += delta;
if (pRx->BitSampleCnt >= pRx->SampleRate)
{
// Another data bit has accumulated
pRx->BitSampleCnt -= pRx->SampleRate;
#if RX_VERBOSE
printf("bit: %u\n", pRx->Deltas < 0);
#endif
pRx->Data >>= 1;
pRx->Data |= (pRx->Deltas < 0) << 9;
pRx->Deltas = delta * reAddDelta;
if ((pRx->Data & 0x201) == 0x200)
{
// Start and stop bits have been detected
if (pRx->State == stCarrierDetected)
{
pRx->State = stReceivingData;
}
pRx->Data = (pRx->Data >> 1) & 0xFF;
pRx->pRxGetDataCallBack(pRx, (uint8)pRx->Data);
#if RX_VERBOSE
printf("byte: 0x%02X ('%c')\n",
pRx->Data,
(pRx->Data >= 0x20 && pRx->Data <= 0x7F) ? pRx->Data : '?');
#endif
pRx->Data = 0x3FF;
}
}
pRx->BitSampleCnt += pRx->BitRate;
}
pRx->LastAngle = angle;
pRx->LastDelta = delta;
pRx->SampleCnt++;
}
typedef struct
{
tTx Tx;
FILE* DataFile;
int CountDown;
} tTxTest;
uint TxGetDataCallBack(tTx* pTx, uint8* pTxData)
{
tTxTest* pTxTest = (tTxTest*)pTx;
uchar c;
if (pTxTest->CountDown)
{
pTxTest->CountDown--;
return 0;
}
if (fread(&c, 1, 1, pTxTest->DataFile) != 1)
{
pTxTest->CountDown = 20;
return 0;
}
*pTxData = c;
return 1;
}
int testTx(uint SampleRate,
double NoiseLevel,
const char* DataFileName,
const char* AudioFileName)
{
FILE *fData = NULL, *fAudio = NULL;
int err = EXIT_FAILURE;
tTxTest txTest;
if ((fData = fopen(DataFileName, "rb")) == NULL)
{
printf("Can't open file \"%s\"\n", DataFileName);
goto Exit;
}
if ((fAudio = fopen(AudioFileName, "wb")) == NULL)
{
printf("Can't create file \"%s\"\n", AudioFileName);
goto Exit;
}
txTest.DataFile = fData;
txTest.CountDown = 0;
TxInit(&txTest.Tx,
SampleRate,
&TxGetDataCallBack);
do
{
int16 sample = TxGetSample(&txTest.Tx);
if (txTest.CountDown > 1 && txTest.CountDown <= 10)
{
#if 0 // Enable this code to test disconnecting.
// Finish with silence.
sample = 0;
#endif
}
sample += (rand() - (int)RAND_MAX / 2) * NoiseLevel * 16000 / (RAND_MAX / 2);
fwrite(&sample, 1, sizeof(sample), fAudio);
} while (txTest.CountDown != 1); // Drain all data-containing samples
err = EXIT_SUCCESS;
Exit:
if (fData != NULL) fclose(fData);
if (fAudio != NULL) fclose(fAudio);
return err;
}
typedef struct
{
tRx Rx;
FILE* DataFile;
} tRxTest;
void RxGetDataCallBack(tRx* pRx, uint8 RxData)
{
tRxTest* pRxTest = (tRxTest*)pRx;
uchar c = RxData;
fwrite(&c, 1, 1, pRxTest->DataFile);
}
int testRx(uint SampleRate,
const char* AudioFileName,
const char* DataFileName)
{
uint lastState;
FILE *fAudio = NULL, *fData = NULL;
int err = EXIT_FAILURE;
tRxTest rxTest;
if ((fAudio = fopen(AudioFileName, "rb")) == NULL)
{
printf("Can't open file \"%s\"\n", AudioFileName);
goto Exit;
}
if ((fData = fopen(DataFileName, "wb")) == NULL)
{
printf("Can't create file \"%s\"\n", DataFileName);
goto Exit;
}
rxTest.DataFile = fData;
RxInit(&rxTest.Rx,
SampleRate,
&RxGetDataCallBack);
for (;;)
{
int16 sample;
if (fread(&sample, 1, sizeof(sample), fAudio) != sizeof(sample))
{
if (rxTest.Rx.State != stCarrierLost) goto NoCarrier;
break;
}
lastState = rxTest.Rx.State;
RxGetSample(&rxTest.Rx, sample);
if (rxTest.Rx.State != lastState && rxTest.Rx.State == stCarrierDetected)
{
printf("\nCONNECT %u\n\n", rxTest.Rx.BitRate);
}
if (rxTest.Rx.State != lastState && rxTest.Rx.State == stCarrierLost)
{
NoCarrier:
printf("\n\nNO CARRIER\n");
break;
}
}
err = EXIT_SUCCESS;
Exit:
if (fAudio != NULL) fclose(fAudio);
if (fData != NULL) fclose(fData);
return err;
}
int main(int argc, char* argv[])
{
uint sampleRate;
double noiseLevel;
if (argc < 2 ||
!stricmp(argv[1], "-help") ||
!stricmp(argv[1], "/help") ||
!stricmp(argv[1], "-?") ||
!stricmp(argv[1], "/?"))
{
Usage:
printf("Usage:\n\n"
" %s tx <sample rate> <noise level> <data input file> <PCM output file>\n"
" %s rx <sample rate> <PCM input file> <data output file>\n",
argv[0],
argv[0]);
return 0;
}
if (!stricmp(argv[1], "tx") &&
argc == 6 &&
sscanf(argv[2], "%u", &sampleRate) == 1 &&
sscanf(argv[3], "%lf", &noiseLevel) == 1)
{
return testTx(sampleRate, noiseLevel, argv[4], argv[5]);
}
else if (!stricmp(argv[1], "rx") &&
argc == 5 &&
sscanf(argv[2], "%u", &sampleRate) == 1)
{
return testRx(sampleRate, argv[3], argv[4]);
}
else
{
goto Usage;
}
}
Typical usage:
modem.exe tx 8000 0.2 testin.txt test8000.pcm
modem.exe rx 8000 test8000.pcm testout.txt
The resulting testout.txt should be identical to testin.txt.
A web search will turn up lots of amateur radio BPSK and RTTY/FSK solutions. Much of this code was written for older slower CPUs, so should run just fine on an iPhone. You can use the Audio Queue API or the RemoteIO Audio Unit for iOS audio IO to the codec.
If you're still looking for a soft modem, you might consider libquiet or Quiet.js. These offer a low power GMSK mode which is fairly capable as well as higher bitrate modes if you're using an audio cable. Quiet uses an existing SDR library to perform its modulation so you'll get something pretty full-featured.
Good day! how сan i use float samples for echo cancellation processing? I tried to change interface and body of central function:
from
void speex_echo_cancellation(SpeexEchoState *st, const spx_int16_t *rec, const spx_int16_t *play, spx_int16_t *out);
to
void float_speex_echo_cancellation(SpeexEchoState *st, const float rec[], const float play[], float out[]);
and from
...
for (i=0;i<st->frame_size;i++)
{
spx_word32_t tmp_out;
tmp_out = SUB32(EXTEND32(st->input[chan*st->frame_size+i]), EXTEND32(st->e[chan*N+i+st->frame_size]));
tmp_out = ADD32(tmp_out, EXTEND32(MULT16_16_P15(st->preemph, st->memE[chan])));
if (in[i*C+chan] <= -32000 || in[i*C+chan] >= 32000)
{
if (st->saturated == 0)
st->saturated = 1;
}
**out[i*C+chan] = (spx_int16_t)WORD2INT(tmp_out);**
st->memE[chan] = tmp_out;
}
...
to
...
for (i=0;i<st->frame_size;i++)
{
spx_word32_t tmp_out;
tmp_out = SUB32(EXTEND32(st->input[chan*st->frame_size+i]), EXTEND32(st->e[chan*N+i+st->frame_size]));
tmp_out = ADD32(tmp_out, EXTEND32(MULT16_16_P15(st->preemph, st->memE[chan])));
if (in[i*C+chan] <= -32000 || in[i*C+chan] >= 32000)
{
if (st->saturated == 0)
st->saturated = 1;
}
**out[i*C+chan] = /*(spx_int16_t)WORD2INT(*/tmp_out*/)*/;**
st->memE[chan] = tmp_out;
}
...
and from
static inline void filter_dc_notch16(const spx_int16_t *in, spx_word16_t radius, spx_word16_t *out, int len, spx_mem_t *mem, int stride)
{
int i;
spx_word16_t den2;
den2 = (spx_word16_t)(radius*radius + .7f*(1.f-radius)*(1.f-radius));
for (i=0;i<len;i++)
{
spx_int16_t vin = in[i*stride];
spx_word32_t vout = mem[0] + SHL32(EXTEND32(vin),15);
mem[0] = mem[1] + 2*(-vin + radius*vout);
mem[1] = SHL32(EXTEND32(vin),15) - MULT16_32_Q15(den2,vout);
out[i] = SATURATE32(PSHR32(MULT16_32_Q15(radius,vout),15),32767);
}
}
to
static inline void float_filter_dc_notch16(const /*spx_int16_t*/spx_word16_t *in, spx_word16_t radius, spx_word16_t *out, int len, spx_mem_t *mem, int stride)
{
int i;
spx_word16_t den2;
den2 = /*(spx_word16_t)*/(radius*radius + .7f*(1.f-radius)*(1.f-radius));
for (i=0;i<len;i++)
{
/*spx_int16_t*/spx_word16_t vin = in[i*stride];
spx_word32_t vout = mem[0] + SHL32(EXTEND32(vin),15);
mem[0] = mem[1] + 2*(-vin + radius*vout);
mem[1] = SHL32(EXTEND32(vin),15) - MULT16_32_Q15(den2,vout);
out[i] = /*SATURATE32(*/PSHR32(MULT16_32_Q15(radius,vout),15)/*,32767)*/;
}
}
So, i prevented conversion from float type output result to short int, but now i get a warning:
speex_warning("The echo canceller started acting funny and got slapped (reset). It swears it will behave now.");
that points to st->screwed_up parameter having 50 values and it signs of setting to zero all out samples:
...
if (!(Syy>=0 && Sxx>=0 && See >= 0)
|| !(Sff < N*1e9 && Syy < N*1e9 && Sxx < N*1e9)
)
{ st->screwed_up += 50; for (i=0;iframe_size*C;i++) out[i] = 0; }
...
What can i do?
enter code here
Why do you want to use float samples?
Standard linear PCM audio is represented as integer samples according to the chosen bitrate - 8 bit, 16 bit and so on.
Where do you get that input from?
If I were you I would just convert whatever you got to shorts and provide it to Speex so it can work with it.