I am working on a routing mechanism that uses information from the MAC layer in Contiki. Therefore, I need to read variables in the csma.c file. So, my question is how is the best way to access variables in this file?
For example, I want to take statistics about the average delay when scheduling a transmission. To that end, I have to read the variable delay in the function schedule_transmission. Specifically, how do I read the variable delay from my routing source file.
static void
schedule_transmission(struct neighbor_queue *n)
{
clock_time_t delay;
int backoff_exponent; /* BE in IEEE 802.15.4 */
backoff_exponent = MIN(n->collisions, CSMA_MAX_BE);
/* Compute max delay as per IEEE 802.15.4: 2^BE-1 backoff periods */
delay = ((1 << backoff_exponent) - 1) * backoff_period();
if(delay > 0) {
/* Pick a time for next transmission */
delay = random_rand() % delay;
}
PRINTF("csma: scheduling transmission in %u ticks, NB=%u, BE=%u\n",
(unsigned)delay, n->collisions, backoff_exponent);
ctimer_set(&n->transmit_timer, delay, transmit_packet_list, n);
}
I think, first you need declare a delay variable before main and second create a function like this in your csma.c:
Clock_time_t get_delay(void)
{
return delay;
}
Then call get_delay() function in your routing source file.
Related
I have an ESP8266 NodeMCU 12E development board and I'm using the Arduino IDE. I'm trying to use a Ticker.h to sample an analog input consistently at a frequency of 10khz, which is one sample every 100us. I noticed that Ticker sampler; sampler.attach(0.0001,callbackfunc); didn't work because attach() won't take the value 0.0001.
So then I wrote the following code based on some guides that I saw:
#include <ESP8266WiFi.h>
#include <Ticker.h>
bool s = true;
void getSample()
{
s = !s;
}
Ticker tickerObject(getSample, 100, 0, MICROS_MICROS);
const char *ssid = "___"; // Change it
const char *pass = "___"; // Change it
void setup()
{
Serial.begin(115200);
Serial.println(0); //start
WiFi.mode(WIFI_STA);
WiFi.begin(ssid, pass);
tickerObject.start();
}
void loop()
{
if(s == true)
{
Serial.println("True");
}
else
{
Serial.println("False");
}
}
However, this did not compile because tickerObject.start() method did not exist. So what I did next was:
Download the latest ticker package as a zip file
Unzip the package from point 1
Made a back up of C:\Users\john\Documents\ArduinoData\packages\esp8266\hardware\esp8266\2.5.0-beta2\libraries\Ticker
Replaced the folder mentioned in point 3 with the Ticker folder in point 2.
Restarted my Arduino IDE
Compiled and ran the code
Opened up the Serial Monitor
However, when I inspect the serial monitor, all it prints is "True". I was expecting the value s to toggle between true and false at a 10khz frequency.
What did I do wrong?
From the documentation of this library:
The library use no interupts of the hardware timers and works with the micros() / millis() function.
This library implements timers in software by polling the micros() and millis() functions. It requires the update() method to be called in loop().
So the start of loop() should be:
void loop()
{
tickerObject.update();
if(s == true)
I'm trying to use a Ticker.h to sample an analog input consistently at a frequency of 10khz
It is worth a go but this is a software based solution that is prone to jitter depending on how often the event loop can be called.
I am using a clone of Arduino UNO, Dccduino and I have problem with the memory.Sketch uses 25,114 bytes (77%) of program storage space. Maximum is 32,256 bytes. Global variables use 1,968 bytes (96%) of dynamic memory, leaving 80 bytes for local variables. Maximum is 2,048 bytes. Low memory available, stability problems may occur.
Is there any way to reduce the memory about 20% if not I think I have to buy Arduino Mega
Here is the code:
#include <OneWire.h>
#include <DallasTemperature.h>
#include <SPI.h>
#include <SD.h>
#include <Wire.h>
#include "RTClib.h"
#include <GPRS_Shield_Arduino.h>
#include <SoftwareSerial.h>
// Data wire is plugged into port 3 and 2 on the Arduino
#define ONE_WIRE_BUS_1 3 // Many sensors on pin 3
#define ONE_WIRE_BUS_2 2 // Many sensors on pin 2
#define TEMPERATURE_PRECISION 9 // Lower resolution
#define PIN_TX 7
#define PIN_RX 8
#define BAUDRATE 9600
#define PHONE_NUMBER "xxxxxxxxxxxxx"
GPRS gprsTest(PIN_TX, PIN_RX, BAUDRATE); //RX,TX,PWR,BaudRate
// Setup a oneWire instance to communicate with any OneWire devices (not just Maxim/Dallas temperature ICs)
OneWire oneWire1(ONE_WIRE_BUS_1);
OneWire oneWire2(ONE_WIRE_BUS_2);
// Pass our oneWire reference to Dallas Temperature.
DallasTemperature sensors1(&oneWire1);
DallasTemperature sensors2(&oneWire2);
int numberOfDevices1; // Number of temperature devices found on pin 3
int numberOfDevices2; // Number of temperature devices found on pin 2
DeviceAddress tempDeviceAddress1; // We'll use this variable to store a found device address for bus 3
DeviceAddress tempDeviceAddress2; // We'll use this variable to store a found device address for bus 2
File myFile;
RTC_DS3231 rtc; // Create a RealTimeClock object
void setup(void)
{
// start serial port
#ifndef ESP8266
while (!Serial); // for Leonardo/Micro/Zero
#endif
Serial.begin(9600);
delay(3000);
Serial.println(F("Dallas Temperature IC Control Library Demo"));
Serial.print( F("Initializing SD card..."));
if (!SD.begin(4)) {
Serial.println(F("\ninitialization failed!"));
return;
}
Serial.println(F("initialization done."));
if (! rtc.begin()) {
Serial.println("Couldn't find RTC");
while (1);
}
if (rtc.lostPower()) {
Serial.println("RTC lost power, lets set the time!");
// following line sets the RTC to the date & time this sketch was compiled
//rtc.adjust(DateTime(F(__DATE__), F(__TIME__)));
// This line sets the RTC with an explicit date & time, for example to set
// January 21, 2014 at 3am you would call:
// rtc.adjust(DateTime(2014, 1, 21, 3, 0, 0));
}
/* while(!gprsTest.init()) { //gprs init
delay(1000);
Serial.print(F("init error\r\n"));
Serial.println(F("gprs init success"));*/ It takes 20% of dynamic memory so i cant use it
}
Serial.println(F("start to call ..."));// Call when device will start
gprsTest.callUp(PHONE_NUMBER);
Serial.println("start to send message ...");
gprsTest.sendSMS(PHONE_NUMBER, "Hi device is ON"); //define phone number and text
// Start up the library
sensors1.begin();
sensors2.begin();
// Grab a count of devices on the wire
numberOfDevices1 = sensors1.getDeviceCount();
numberOfDevices2 = sensors2.getDeviceCount();
// locate devices on the bus
Serial.print(F("Locating devices..."));
Serial.print(F("Found "));
Serial.print(numberOfDevices1, DEC );
Serial.print(F("+"));
Serial.print(numberOfDevices2, DEC );
Serial.println(F(" devices."));
// report parasite power requirements
Serial.print("Parasite power is: ");
if (sensors1.isParasitePowerMode()) Serial.println(F("Sensors 1 ON"));
else Serial.println(F("\nSensors 1 OFF"));
if (sensors2.isParasitePowerMode()) Serial.println(F("Sensors 2 ON"));
else Serial.println(F("Sensors 2 OFF"));
// Loop through each device, print out address for pin 3
for (int i = 0; i < numberOfDevices1; i++)
{
// Search the wire for address
if (sensors1.getAddress(tempDeviceAddress1, i))
{
Serial.print(F("Found device "));
Serial.print(i, DEC);
Serial.print(F(" with address: "));
printAddress(tempDeviceAddress1);
Serial.println();
Serial.println(F("\n"));
// set the resolution to TEMPERATURE_PRECISION bit (Each Dallas/Maxim device is capable of several different resolutions)
sensors1.setResolution(tempDeviceAddress1, TEMPERATURE_PRECISION);
} else {
Serial.print(F("Found ghost device for pin 3 at "));
Serial.print(i, DEC);
Serial.print(F(" but could not detect address. Check power and cabling"));
}
}
// Loop through each device, print out address for pin 2
for (int i = 0; i < numberOfDevices2; i++)
{
// Search the wire for address
if (sensors2.getAddress(tempDeviceAddress2, i))
{
Serial.print(F("Found device "));
Serial.print(i + numberOfDevices1, DEC);
Serial.print(F(" with address: "));
printAddress(tempDeviceAddress2);
Serial.println();
Serial.println(F("\n"));
// set the resolution to TEMPERATURE_PRECISION bit (Each Dallas/Maxim device is capable of several different resolutions)
sensors2.setResolution(tempDeviceAddress2, TEMPERATURE_PRECISION);
} else {
Serial.print(F("Found ghost device for pin 2 at "));
Serial.print(i, DEC);
Serial.print(F(" but could not detect address. Check power and cabling"));
}
}
}
void loop(void)
{
// call sensors1.requestTemperatures() to issue a global temperature
// request to all devices on the bus
Serial.print(F("Requesting temperatures to pin 3..."));
sensors1.requestTemperatures(); // Send the command to get temperatures for pin 3
Serial.println(F("DONE"));
myFile = SD.open("test1.txt", FILE_WRITE); //open file
// Loop through each device , print out temperature data for pin 3
for (int i = 0; i < numberOfDevices1; i++)
{
// Search the wire for address
if (sensors1.getAddress(tempDeviceAddress1, i))
{
// Output the device ID
Serial.print(F("Temperature for device: "));
Serial.println(i, DEC);
// It responds almost immediately. Let's print out the data
printTemperature1(tempDeviceAddress1);// Use a simple function to print out the data
Serial.print(F("\n"));
}
delay(4000);
//else ghost device! Check your power requirements and cabling
}// End forloop for pin 3
if (numberOfDevices2 != 0) {
Serial.print(F("Requesting temperatures to pin 2..."));
sensors2.requestTemperatures(); // Send the command to get temperatures for pin 2
Serial.println(F("DONE"));
}
// Loop through each device for pin 2, print out temperature data
for (int i = 0; i < numberOfDevices2; i++)
{
// Search the wire for address
if (sensors2.getAddress(tempDeviceAddress2, i))
{
// Output the device ID
Serial.print(F("Temperature for device: "));
Serial.println(i + numberOfDevices1, DEC);
// It responds almost immediately. Let's print out the data
printTemperature2(tempDeviceAddress2);// Use a simple function to print out the data
Serial.print(F("\n"));
}
else Serial.print(F("ghost device! Check your power requirements and cabling"));
delay(4000);
} //End forloop for pin 3
myFile.close(); // Should I close it?
}// End loop()
void printAddress(DeviceAddress deviceAddress) // function to print a device address
{
for (uint8_t i = 0; i < 8; i++)
{
if (deviceAddress[i] < 16) Serial.print(F("0"));
Serial.print(deviceAddress[i], HEX);
}
}
void printTemperature1(DeviceAddress deviceAddress1) // function to print the temperature for a device (pin 3)
{
float tempC = sensors1.getTempC(deviceAddress1);
Serial.print("Temp C: ");
Serial.print(tempC);
if (myFile)
{
Serial.println(F("\nWriting to test.txt..."));
myFile.print(F("C: "));
myFile.print(tempC);
print_time(); // Call print_time() function to print time on file
myFile.print(F("\n"));
Serial.print(F("Done!"));
}
else Serial.print(F("Error opening file 1"));
Serial.println("\n");
}
void printTemperature2(DeviceAddress deviceAddress2) // function to print the temperature for a device (pin 2)
{
float tempC = sensors2.getTempC(deviceAddress2);
Serial.print(F("Temp C: "));
Serial.print(tempC);
if (myFile)
{
Serial.print(F("\nWriting to test.txt..."));
myFile.print(F("C: "));
myFile.print(tempC);
print_time(); // Call print_time() function to print time on file
myFile.print(F("\n"));
Serial.print(F("Done!"));
} else Serial.print(F("Error opening file 2"));
Serial.println("\n");
}
void print_time() { // print time function
DateTime now = rtc.now();
Serial.print(now.year(), DEC);
Serial.print('/');
Serial.print(now.month(), DEC);
Serial.print('/');
Serial.print(now.day(), DEC);
Serial.print(now.hour(), DEC);
Serial.print(':');
Serial.print(now.minute(), DEC);
Serial.print(':');
Serial.print(now.second(), DEC);
Serial.println();
myFile.print(now.year(), DEC);
myFile.print('/');
myFile.print(now.month(), DEC);
myFile.print('/');
myFile.print(now.day(), DEC);
myFile.print(now.hour(), DEC);
myFile.print(':');
myFile.print(now.minute(), DEC);
myFile.print(':');
myFile.print(now.second(), DEC);
myFile.println();
}
I'm doing more than that for a system in my barn - SD card, RTC, LCD display, GPRS modem, radio communication with other devices, controlling pumps based on programmed timing by season, rain sensors, float sensors, temp sensors, voltage sensors, etc. Here are some things I've found:
All string literals should be replaced by F() macro calls and or native string functions working on flash. So, strcpy_P(string1, PSTR(string2)), strcat_P(string1, PSTR(string2)) kind of calls.
Turn lots of accesses to devices into function calls that contain any data structures needed for those calls. Your arduino will work harder adding and removing stack and stack frames, but the data structures will be removed from the stack when the function is done, and machine cycles are much cheaper on Arduino than RAM. So, separate your temperature reading code in a separate function from your file-writing. Return the float, then send that float to your SD-writing function.
Hide all of your Serial code in debug_print code instead. So, you'll use debug_print calls when debugging, and they all disappear completely from your production code.
Make sure you're only calling the SD code in subroutines, and instantiating the actual FAT code in those subroutines as needed, not as a big global.
There are a number of different SD libraries, some are cheaper (memory-wise) than others. Shop around.
Use one of the freeMemory variants to decide if you have the memory available before calling one of those RAM-hungry SD calls. If you don't have enough RAM at the time, you may want to implement some kind of circular buffer in EEPROM to store messages to be written to SD when you do have the RAM available.
Use booleans and bytes instead of ints when possible, and consider using bit fields for flags to save even more RAM. Are you really going to have up to 32,000 devices on your temperature buses? You can get 255 in a byte.
I think I've seen this code before on another site.
From your code I assume you're making a temperature logger and you want to log the data on the SD card.
Most of your code is redundant if you use the DS1307RTC.h library and the Time.h library. DS1307RTC is a generic RTC library. With it you don't need OneWire, Wire or SPI and Wire. Software Serial also is unnecessary.
However, I suggest you check out my Arduino DataLogger Library on Github: https://github.com/FreelanceJavaDev/DataLogger
I've pretty much maxed out the memory I saved on my Uno down to 22,680 bytes (70%) of program storage space and 1,237 bytes (60%) of SRAM (Dynamic memory). It automatically configures the RTC and SD card. It makes a CSV file for export to excel for each month it's running organized by date.
I've built several Arduino data loggers in my time, and all of the ones using SD cards have failed because of lack of memory. The SD on its own uses half of the memory available on an ATMega 328. Add to that a couple of libraries for other hardware and you have no memory left at all for your sketch.
I have gone over to 24LC512s. One is normally enough, but you can use up to 4 with different addresses if you want. This is a relatively small amount of memory, but I have found it is always enough. It is too easy to generate megabytes of data which is too large to be analysed. One 24LC512 holds more than enough data for one spreadsheet. The only down side is that you have to use an Arduino to read back the data through the USB.
I use the first two bytes to store the number of records, and the next byte to store the number of bytes per record. (Although in retrospect the latter is not really necessary.) You might think that the first two bytes would “wear out” because they are re-written every time a new entry is made, but this has not happened to me yet. I have had the same Arduino continuously running for 7 years now (uninterrupted apart from a few updates to the sketch), and generating over 1000 records per month, so the first two bytes must have been updated that number of times. I have never had a problem with the first two bytes, and even if I did, it would be cheap enough to replace the 24LC512.
You can even get away with “hot plugging”: I leave the logger running and change over the memory chip for a fresh one so that I can read the data without interrupting the logger. (Read the number of records, then increment it and write the new number and the data.)
I am using Audio Queues to playback audio files. I need precise timing on the finish of last buffer.
I need to notify a function no later than 150ms-200 ms after the last buffer is played...
Thru callback method I know how many buffers are enqueued
I know the buffer size, I know the how many bytes last buffer is filled with.
First I initialize a number of buffers end fill the buffers with audio data, then enqueue them. When Audio Queue needs a buffer to be filled it calls the callback and I fill the buffer with data.
When there is no more audio data available Audio Queue sends me the last empty buffer, so I fill it with whatever data I have:
if (sharedCache.numberOfToTalPackets>0)
{
if (currentlyReadingBufferIndex==[sharedCache.baseAudioCache count]-1) {
inBuffer->mAudioDataByteSize = (UInt32)bytesFilled;
lastEnqueudBufferSize=bytesFilled;
err=AudioQueueEnqueueBuffer(inAQ,inBuffer,(UInt32)packetsFilled,packetDescs);
if (err) {
[self failWithErrorCode:err customError:AP_AUDIO_QUEUE_ENQUEUE_FAILED];
}
printf("if that was the last free packet description, then enqueue the buffer\n");
//go to the next item on keepbuffer array
isBufferFilled=YES;
[self incrementBufferUsedCount];
return;
}
}
When Audio Queue asks for more data via callback and I have no more data , I start to countdown the buffers. If buffer count equals to zero, which means only one buffer left on the flight to be played, the moment playback is done I try to stop the audio queue.
-(void)decrementBufferUsedCount
{
if (buffersUsed>0) {
buffersUsed--;
printf("buffer on the queue %i\n",buffersUsed);
if (buffersUsed==0) {
NSLog(#"playback is finished\n");
// end playback
isPlayBackDone=YES;
double sampleRate = dataFormat.mSampleRate;
double bufferDuration = lastEnqueudBufferSize/ sampleRate;
double estimatedTimeNeded=bufferDuration*1;
[self performSelector:#selector(stopPlayer) withObject:nil afterDelay:estimatedTimeNeded];
}
}
}
-(void)stopPlayer
{
#synchronized(self)
{
state=AP_STOPPING;
}
err=AudioQueueStop(queue, TRUE);
if (err) {
[self failWithErrorCode:err customError:AP_AUDIO_QUEUE_STOP_FAILED];
}
else
{
#synchronized(self)
{
state=AP_STOPPED;
NSLog(#"Stopped\n");
}
However it seems I can't get precise timing here. Above code stops player early.
if I do following audio cuts early too
double bufferDuration = XMAQDefaultBufSize/ sampleRate;
double estimatedTimeNeded=bufferDuration*1;
if increase 1 to 2 since the buffer size is big I get some delay, seem 1.5 is the optimum value for now but I dont understand why lastEnqueudBufferSize/ sampleRate is not wotking
Details of the audio file, and buffers:
Audio file has 22050 sample rate
#define kNumberPlaybackBuffers 4
#define kAQDefaultBufSize 16384
it is a vbr file format with no bitrate information available
EDIT:
I found an easier way that gets the same results (+/-10ms). After you set up your output Queue with AudioQueueNewOutput() you initialize a AudioQueueTimelineRef to be used in your output callback. (ticksToSeconds function is included below in my first method) don't forget to import<mach/mach_time.h>
//After AudioQueueNewOutput()
AudioQueueTimelineRef timeLine; //ivar
AudioQueueCreateTimeline(queue, self.timeLine);
Then in your output callback you call AudioQueueGetCurrentTime(). Caveat: queue must be playing for valid timestamps. So for very short files you might need to use the AudioQueueProcessingTap method below.
AudioTimeStamp timestamp;
AudioQueueGetCurrentTime(queue, self->timeLine, ×tamp, NULL);
The timestamp ties together the current sample playing with the current machine time. With that info we can get an exact machine time in the future when our last sample will be played.
Float64 samplesLeft = self->frameCount - timestamp.mSampleTime;//samples in file - current sample
Float64 secondsLeft = samplesLeft / self->sampleRate; //seconds of audio to play
UInt64 ticksLeft = secondsLeft / ticksToSeconds(); //seconds converted to machine ticks
UInt64 machTimeFinish = timestamp.mHostTime + ticksLeft; //machine time of first sample + ticks left
Now that we have this future machine time we can use it to time whatever it is that you want to do with some accuracy.
UInt64 currentMachTime = mach_absolute_time();
Uint64 ticksFromNow = machTimeFinish - currentMachTime;
float secondsFromNow = ticksFromNow * ticksToSeconds();
dispatch_after(dispatch_time(DISPATCH_TIME_NOW, (int64_t)(secondsFromNow * NSEC_PER_SEC)), dispatch_get_main_queue(), ^{
//do the thing!!!
printf("Giggety");
});
If GCD dispatch_async isn't accurate enough there are ways to set up a precision timer
Using AudioQueueProcessingTap
You can get fairly low response time from an AudioQueueProcessingTap. First you make your callback that will essentially put itself in-between the audio stream. The MyObject type is just whatever self is in your code(this is ARC bridging here to get self inside the function). Inspecting ioFlags tells you when the stream starts and finishes. The ioTimeStamp of an output callback describes time that the first sample in the callback will hit the speaker in the future. So if you want to get exact here's how you do it. I added some convenience functions for converting machine time to seconds.
#import <mach/mach_time.h>
double getTimeConversion(){
double timecon;
mach_timebase_info_data_t tinfo;
kern_return_t kerror;
kerror = mach_timebase_info(&tinfo);
timecon = (double)tinfo.numer / (double)tinfo.denom;
return timecon;
}
double ticksToSeconds(){
static double ticksToSeconds = 0;
if (!ticksToSeconds) {
ticksToSeconds = getTimeConversion() * 0.000000001;
}
return ticksToSeconds;
}
void processingTapCallback(
void * inClientData,
AudioQueueProcessingTapRef inAQTap,
UInt32 inNumberFrames,
AudioTimeStamp * ioTimeStamp,
UInt32 * ioFlags,
UInt32 * outNumberFrames,
AudioBufferList * ioData){
MyObject *self = (__bridge Object *)inClientData;
AudioQueueProcessingTapGetSourceAudio(inAQTap, inNumberFrames, ioTimeStamp, ioFlags, outNumberFrames, ioData);
if (*ioFlags == kAudioQueueProcessingTap_EndOfStream) {
Float64 sampTime;
UInt32 frameCount;
AudioQueueProcessingTapGetQueueTime(inAQTap, &sampTime, &frameCount);
Float64 samplesInThisCallback = self->frameCount - sampleTime;//file sampleCount - queue current sample
//double secondsInCallback = outNumberFrames / (double)self->sampleRate; outNumberFrames was inaccurate
double secondsInCallback = * samplesInThisCallback / (double)self->sampleRate;
uint64_t timeOfLastSampleLeavingSpeaker = ioTimeStamp->mHostTime + (secondsInCallback / ticksToSeconds());
[self lastSampleDoneAt:timeOfLastSampleLeavingSpeaker];
}
}
-(void)lastSampleDoneAt:(uint64_t)lastSampTime{
uint64_t currentTime = mach_absolute_time();
if (lastSampTime > currentTime) {
double secondsFromNow = (lastSampTime - currentTime) * ticksToSeconds();
dispatch_after(dispatch_time(DISPATCH_TIME_NOW, (int64_t)(secondsFromNow * NSEC_PER_SEC)), dispatch_get_main_queue(), ^{
//do the thing!!!
});
}
else{
//do the thing!!!
}
}
You set it up like this after AudioQueueNewOutput and before AudioQueueStart. Notice the passing of bridged self to the inClientData argument. The queue actually holds self as void* to be used in callback where we bridge it back to an objective-C object within the callback.
AudioStreamBasicDescription format;
AudioQueueProcessingTapRef tapRef;
UInt32 maxFrames = 0;
AudioQueueProcessingTapNew(queue, processingTapCallback, (__bridge void *)self, kAudioQueueProcessingTap_PostEffects, &maxFrames, &format, &tapRef);
You could get the end machine time as soon as the file starts too. A little cleaner too.
void processingTapCallback(
void * inClientData,
AudioQueueProcessingTapRef inAQTap,
UInt32 inNumberFrames,
AudioTimeStamp * ioTimeStamp,
UInt32 * ioFlags,
UInt32 * outNumberFrames,
AudioBufferList * ioData){
MyObject *self = (__bridge Object *)inClientData;
AudioQueueProcessingTapGetSourceAudio(inAQTap, inNumberFrames, ioTimeStamp, ioFlags, outNumberFrames, ioData);
if (*ioFlags == kAudioQueueProcessingTap_StartOfStream) {
uint64_t timeOfLastSampleLeavingSpeaker = ioTimeStamp->mHostTime + (self->audioDurSeconds / ticksToSeconds());
[self lastSampleDoneAt:timeOfLastSampleLeavingSpeaker];
}
}
If you use AudioQueueStop in asynchronous mode, then stopping happens after all queued buffers have been played or recorded. See doc.
You're using it in a synchronous mode, where stopping happens ASAP, and playback cuts out immediately, without regard for previously buffered audio data. You want precise timing, but only because audio is cutting off. Right? So rather than go synchronous + add additional timing/callback code, I recommend going asynchronous:
err=AudioQueueStop(queue, FALSE);
From docs:
If you pass false, the function returns immediately, but the audio
queue does not stop until its queued buffers are played or recorded
(that is, the stop occurs asynchronously). Audio queue callbacks are
invoked as necessary until the queue actually stops.
For me this worked really well for what I heeded:
stopping the queue in callback when data is over using AudioQueueStop(queue, FALSE), while:
listening to actual stop using kAudioQueueProperty_IsRunning property (happens later than AudioQueueStop() is called, actually, when last buffer gets actually rendered)
after stopping the queue You can get prepared for action You need to execute on audio ending, and when listener fires - actually execute this action.
I am not sure about time precision of that event but for my task it behaved definitely better than using notification straight from callback. There is buffering inside AudioQueue and output device itself so definitely IsRunning listener gives better results as to when AudioQueue stops playing.
I try to implement following logic (a kind of pseudo-code) using pthread:
pthread_mutex_t mutex;
threadA()
{
lock(mutex);
// do work
timed_lock(mutex, current_abs_time + 1 minute);
}
threadB()
{
// do work in more than 1 minute
unlock(mutex);
}
I do expect threadA to do the work and wait untill threadB signals but not longer than 1 minute. I have done similar a lot of time in Win32 but stuck with pthreads: a timed_lock part returns imediately (not in 1 minute) with code ETIMEDOUT.
Is there a simple way to implement the logic above?
even following code returns ETIMEDOUT immediately
pthread_mutex_t m;
// Thread A
pthread_mutex_init(&m, 0);
pthread_mutex_lock(&m);
// Thread B
struct timespec now;
clock_gettime(CLOCK_MONOTONIC, &now);
struct timespec time = {now.tv_sec + 5, now.tv_nsec};
pthread_mutex_timedlock(&m, &time); // immediately return ETIMEDOUT
Does anyone know why? I have also tried with gettimeofday function
Thanks
I implemented my logic with conditional variables with respect to other rules (using wrapping mutex, bool flag etc.)
Thank you all for comments.
For the second piece of code: AFAIK pthread_mutex_timedlock only works with CLOCK_REALTIME.
CLOCK_REALTIME are seconds since 01/01/1970
CLOCK_MONOTONIC typically since boot
Under these premises, the timeout set is few seconds into 1970 and therefore in the past.
try something like this :
class CmyClass
{
boost::mutex mtxEventWait;
bool WaitForEvent(long milliseconds);
boost::condition cndSignalEvent;
};
bool CmyClass::WaitForEvent(long milliseconds)
{
boost::mutex::scoped_lock mtxWaitLock(mtxEventWait);
boost::posix_time::time_duration wait_duration = boost::posix_time::milliseconds(milliseconds);
boost::system_time const timeout=boost::get_system_time()+wait_duration;
return cndSignalEvent.timed_wait(mtxEventWait,timeout); // wait until signal Event
}
// so inorder to wait then call the WaitForEvent method
WaitForEvent(1000); // it will timeout after 1 second
// this is how an event could be signaled:
cndSignalEvent.notify_one();
I'm looking to build an incredibly simple application for iOS with a button that starts and stops an audio signal. The signal is just going to be a sine wave, and it's going to check my model (an instance variable for the volume) throughout its playback and change its volume accordingly.
My difficulty has to do with the indefinite nature of the task. I understand how to build tables, fill them with data, respond to button presses, and so on; however, when it comes to just having something continue on indefinitely (in this case, a sound), I'm a little stuck! Any pointers would be terrific!
Thanks for reading.
Here's a bare-bones application which will play a generated frequency on-demand. You haven't specified whether to do iOS or OSX, so I've gone for OSX since it's slightly simpler (no messing with Audio Session Categories). If you need iOS, you'll be able to find out the missing bits by looking into Audio Session Category basics and swapping the Default Output audio unit for the RemoteIO audio unit.
Note that the intention of this is purely to demonstrate some Core Audio / Audio Unit basics. You'll probably want to look into the AUGraph API if you want to start getting more complex than this (also in the interest of providing a clean example, I'm not doing any error checking. Always do error checking when dealing with Core Audio).
You'll need to add the AudioToolbox and AudioUnit frameworks to your project to use this code.
#import <AudioToolbox/AudioToolbox.h>
#interface SWAppDelegate : NSObject <NSApplicationDelegate>
{
AudioUnit outputUnit;
double renderPhase;
}
#end
#implementation SWAppDelegate
- (void)applicationDidFinishLaunching:(NSNotification *)aNotification
{
// First, we need to establish which Audio Unit we want.
// We start with its description, which is:
AudioComponentDescription outputUnitDescription = {
.componentType = kAudioUnitType_Output,
.componentSubType = kAudioUnitSubType_DefaultOutput,
.componentManufacturer = kAudioUnitManufacturer_Apple
};
// Next, we get the first (and only) component corresponding to that description
AudioComponent outputComponent = AudioComponentFindNext(NULL, &outputUnitDescription);
// Now we can create an instance of that component, which will create an
// instance of the Audio Unit we're looking for (the default output)
AudioComponentInstanceNew(outputComponent, &outputUnit);
AudioUnitInitialize(outputUnit);
// Next we'll tell the output unit what format our generated audio will
// be in. Generally speaking, you'll want to stick to sane formats, since
// the output unit won't accept every single possible stream format.
// Here, we're specifying floating point samples with a sample rate of
// 44100 Hz in mono (i.e. 1 channel)
AudioStreamBasicDescription ASBD = {
.mSampleRate = 44100,
.mFormatID = kAudioFormatLinearPCM,
.mFormatFlags = kAudioFormatFlagsNativeFloatPacked,
.mChannelsPerFrame = 1,
.mFramesPerPacket = 1,
.mBitsPerChannel = sizeof(Float32) * 8,
.mBytesPerPacket = sizeof(Float32),
.mBytesPerFrame = sizeof(Float32)
};
AudioUnitSetProperty(outputUnit,
kAudioUnitProperty_StreamFormat,
kAudioUnitScope_Input,
0,
&ASBD,
sizeof(ASBD));
// Next step is to tell our output unit which function we'd like it
// to call to get audio samples. We'll also pass in a context pointer,
// which can be a pointer to anything you need to maintain state between
// render callbacks. We only need to point to a double which represents
// the current phase of the sine wave we're creating.
AURenderCallbackStruct callbackInfo = {
.inputProc = SineWaveRenderCallback,
.inputProcRefCon = &renderPhase
};
AudioUnitSetProperty(outputUnit,
kAudioUnitProperty_SetRenderCallback,
kAudioUnitScope_Global,
0,
&callbackInfo,
sizeof(callbackInfo));
// Here we're telling the output unit to start requesting audio samples
// from our render callback. This is the line of code that starts actually
// sending audio to your speakers.
AudioOutputUnitStart(outputUnit);
}
// This is our render callback. It will be called very frequently for short
// buffers of audio (512 samples per call on my machine).
OSStatus SineWaveRenderCallback(void * inRefCon,
AudioUnitRenderActionFlags * ioActionFlags,
const AudioTimeStamp * inTimeStamp,
UInt32 inBusNumber,
UInt32 inNumberFrames,
AudioBufferList * ioData)
{
// inRefCon is the context pointer we passed in earlier when setting the render callback
double currentPhase = *((double *)inRefCon);
// ioData is where we're supposed to put the audio samples we've created
Float32 * outputBuffer = (Float32 *)ioData->mBuffers[0].mData;
const double frequency = 440.;
const double phaseStep = (frequency / 44100.) * (M_PI * 2.);
for(int i = 0; i < inNumberFrames; i++) {
outputBuffer[i] = sin(currentPhase);
currentPhase += phaseStep;
}
// If we were doing stereo (or more), this would copy our sine wave samples
// to all of the remaining channels
for(int i = 1; i < ioData->mNumberBuffers; i++) {
memcpy(ioData->mBuffers[i].mData, outputBuffer, ioData->mBuffers[i].mDataByteSize);
}
// writing the current phase back to inRefCon so we can use it on the next call
*((double *)inRefCon) = currentPhase;
return noErr;
}
- (void)applicationWillTerminate:(NSNotification *)notification
{
AudioOutputUnitStop(outputUnit);
AudioUnitUninitialize(outputUnit);
AudioComponentInstanceDispose(outputUnit);
}
#end
You can call AudioOutputUnitStart() and AudioOutputUnitStop() at will to start/stop producing audio. If you want to dynamically change the frequency, you can pass in a pointer to a struct containing both the renderPhase double and another one representing the frequency you want.
Be careful in the render callback. It's called from a realtime thread (not from the same thread as your main run loop). Render callbacks are subject to some fairly strict time requirements, which means that there's many things you Should Not Do in your callback, such as:
Allocate memory
Wait on a mutex
Read from a file on disk
Objective-C messaging (Yes, seriously.)
Note that this is not the only way to do this. I've only demonstrated it this way since you've tagged this core-audio. If you don't need to change the frequency you can just use the AVAudioPlayer with a pre-made sound file containing your sine wave.
There's also Novocaine, which hides a lot of this verbosity from you. You could also look into the Audio Queue API, which works fairly similar to the Core Audio sample I wrote but decouples you from the hardware a little more (i.e. it's less strict about how you behave in your render callback).