I am looking for a way to track iOS memory usage from within my app as accurately as possible. After many trials I end up with the following code:
uint64_t memory_usage (void) {
task_vm_info_data_t vmInfo;
mach_msg_type_number_t count = TASK_VM_INFO_COUNT;
if (task_info(mach_task_self(), TASK_VM_INFO, (task_info_t) &vmInfo, &count) != KERN_SUCCESS) return -1;
return vmInfo.internal;
}
When executed from a real device the value returned is in sync with the value reported by the Xcode debugger.
The behaviour that I cannot explain is when allocated memory reaches about 500MB the value returned by this function do not increase (it continue to report a value in the 450/450MB range) while the Xcode debugger continue to grow up until 1.3GB and then a memory warning is notified to my app.
The code I used to test this function is fired every 0.5 seconds from within an NSTimer with the following action:
- (void) checkMemory {
uint64_t mem_usage = memory_usage();
// allocate a 5MB buffer to test memory usage
uint32_t buffer_size = 5 * 1024 * 1024;
char *buffer = (char *)malloc(buffer_size);
memset(buffer, 0, buffer_size);
p[i++] = buffer;
}
Am I missing something?
Related
I'm using an STM32 (STM32F446RE) to receive audio from two INMP441 mems microphone in an stereo setup via I2S protocol and record it into a .WAV on a micro SD card, using the HAL library.
I wrote the firmware that records audio into a .WAV with FreeRTOS. But the audio files that I record sound like Darth Vader. Here is a screenshot of the audio in audacity:
if you zoom in you can see a constant noise being inserted in between the real audio data:
I don't know what is causing this.
I have tried increasing the MessageQueue, but that doesnt seem to be the problem, the queue is kept at 0 most of the time. I've tried different frame sizes and sampling rates, changing the number of channels, using only one inmp441. All this without any success.
I proceed explaining the firmware.
Here is a block diagram of the architecture for the RTOS that I have implemented:
It consists of three tasks. The first one receives a command via UART (with interrupts) that signals to start or stop recording. the second one is simply an state machine that walks through the steps to write a .WAV.
Here the code for the WriteWavFileTask:
switch(audio_state)
{
case STATE_START_RECORDING:
sprintf(filename, "%saud_%03d.wav", SDPath, count++);
do
{
res = f_open(&file_ptr, filename, FA_CREATE_ALWAYS|FA_WRITE);
}
while(res != FR_OK);
res = fwrite_wav_header(&file_ptr, I2S_SAMPLE_FREQUENCY, I2S_FRAME, 2);
HAL_I2S_Receive_DMA(&hi2s2, aud_buf, READ_SIZE);
audio_state = STATE_RECORDING;
break;
case STATE_RECORDING:
osDelay(50);
break;
case STATE_STOP:
HAL_I2S_DMAStop(&hi2s2);
while(osMessageQueueGetCount(AudioQueueHandle)) osDelay(1000);
filesize = f_size(&file_ptr);
data_len = filesize - 44;
total_len = filesize - 8;
f_lseek(&file_ptr, 4);
f_write(&file_ptr, (uint8_t*)&total_len, 4, bw);
f_lseek(&file_ptr, 40);
f_write(&file_ptr, (uint8_t*)&data_len, 4, bw);
f_close(&file_ptr);
audio_state = STATE_IDLE;
break;
case STATE_IDLE:
osThreadSuspend(WAVHandle);
audio_state = STATE_START_RECORDING;
break;
default:
osDelay(50);
break;
Here are the macros used in the code for readability:
#define I2S_DATA_WORD_LENGTH (24) // industry-standard 24-bit I2S
#define I2S_FRAME (32) // bits per sample
#define READ_SIZE (128) // samples to read from I2S
#define WRITE_SIZE (READ_SIZE*I2S_FRAME/16) // half words to write
#define WRITE_SIZE_BYTES (WRITE_SIZE*2) // bytes to write
#define I2S_SAMPLE_FREQUENCY (16000) // sample frequency
The last task is the responsible for processing the buffer received via I2S. Here is the code:
void convert_endianness(uint32_t *array, uint16_t Size) {
for (int i = 0; i < Size; i++) {
array[i] = __REV(array[i]);
}
}
void HAL_I2S_RxCpltCallback(I2S_HandleTypeDef *hi2s)
{
convert_endianness((uint32_t *)aud_buf, READ_SIZE);
osMessageQueuePut(AudioQueueHandle, aud_buf, 0L, 0);
HAL_I2S_Receive_DMA(hi2s, aud_buf, READ_SIZE);
}
void pvrWriteAudioTask(void *argument)
{
/* USER CODE BEGIN pvrWriteAudioTask */
static UINT *bw;
static uint16_t aud_ptr[WRITE_SIZE];
/* Infinite loop */
for(;;)
{
osMessageQueueGet(AudioQueueHandle, aud_ptr, 0L, osWaitForever);
res = f_write(&file_ptr, aud_ptr, WRITE_SIZE_BYTES, bw);
}
/* USER CODE END pvrWriteAudioTask */
}
This tasks reads from a queue an array of 256 uint16_t elements containing the raw audio data in PCM. f_write takes the Size parameter in number of bytes to write to the SD card, so 512 bytes. The I2S Receives 128 frames (for a 32 bit frame, 128 words).
The following is the configuration for the I2S and clocks:
Any help would be much appreciated!
Solution
As pmacfarlane pointed out, the problem was with the method used for buffering the audio data. The solution consisted of easing the overhead on the ISR and implementing a circular DMA for double buffering. Here is the code:
#define I2S_DATA_WORD_LENGTH (24) // industry-standard 24-bit I2S
#define I2S_FRAME (32) // bits per sample
#define READ_SIZE (128) // samples to read from I2S
#define BUFFER_SIZE (READ_SIZE*I2S_FRAME/16) // number of uint16_t elements expected
#define WRITE_SIZE_BYTES (BUFFER_SIZE*2) // bytes to write
#define I2S_SAMPLE_FREQUENCY (16000) // sample frequency
uint16_t aud_buf[2*BUFFER_SIZE]; // Double buffering
static volatile int16_t *BufPtr;
void convert_endianness(uint32_t *array, uint16_t Size) {
for (int i = 0; i < Size; i++) {
array[i] = __REV(array[i]);
}
}
void HAL_I2S_RxHalfCpltCallback(I2S_HandleTypeDef *hi2s)
{
BufPtr = aud_buf;
osSemaphoreRelease(RxAudioSemHandle);
}
void HAL_I2S_RxCpltCallback(I2S_HandleTypeDef *hi2s)
{
BufPtr = &aud_buf[BUFFER_SIZE];
osSemaphoreRelease(RxAudioSemHandle);
}
void pvrWriteAudioTask(void *argument)
{
/* USER CODE BEGIN pvrWriteAudioTask */
static UINT *bw;
/* Infinite loop */
for(;;)
{
osSemaphoreAcquire(RxAudioSemHandle, osWaitForever);
convert_endianness((uint32_t *)BufPtr, READ_SIZE);
res = f_write(&file_ptr, BufPtr, WRITE_SIZE_BYTES, bw);
}
/* USER CODE END pvrWriteAudioTask */
}
Problems
I think the problem is your method of buffering the audio data - mainly in this function:
void HAL_I2S_RxCpltCallback(I2S_HandleTypeDef *hi2s)
{
convert_endianness((uint32_t *)aud_buf, READ_SIZE);
osMessageQueuePut(AudioQueueHandle, aud_buf, 0L, 0);
HAL_I2S_Receive_DMA(hi2s, aud_buf, READ_SIZE);
}
The main problem is that you are re-using the same buffer each time. You have queued a message to save aud_buf to the SD-card, but you've also instructed the I2S to start DMAing data into that same buffer, before it has been saved. You'll end up saving some kind of mish-mash of "old" data and "new" data.
#Flexz pointed out that the message queue takes a copy of the data, so there is no issue about the I2S writing over the data that is being written to the SD-card. However, taking the copy (in an ISR) adds overhead, and delays the start of the new I2S DMA.
Another problem is that you are doing the endian conversion in this function (that is called from an ISR). This will block any other (lower priority) interrupts from being serviced while this happens, which is a bad thing in an embedded system. You should do the endian conversion in the task that reads from the queue. ISRs should be very short and do the minimum possible work (often just setting a flag, giving a semaphore, or adding something to a queue).
Lastly, while you are doing the endian conversion, what is happening to audio samples? The previous DMA has completed, and you haven't started a new one, so they will just be dropped on the floor.
Possible solution
You probably want to allocate a suitably big buffer, and configure your DMA to work in circular buffer mode. This means that once started, the DMA will continue forever (until you stop it), so you'll never drop any samples. There won't be any gap between one DMA finishing and a new one starting, since you never need to start a new one.
The DMA provides a "half-complete" interrupt, to say when it has filled half the buffer. So start the DMA, and when you get the half-complete interrupt, queue up the first half of the buffer to be saved. When you get the fully-complete interrupt, queue up the second half of the buffer to be saved. Rinse and repeat.
You might want to add some logic to detect if the interrupt happens before the previous save has completed, since the data will be overrun and possibly corrupted. Depending on the speed of the SD-card (and the sample rate), this may or may not be a problem.
I'm having trouble understanding why there are time bubbles on my GPU timeline when inspecting my app using PIX timing captures. Here is a picture of one of the time bubbles I'm talking about, highlighted in orange:
The timeline actually doesn't look at all how I expected. Since I am triple buffering, I would expect the GPU to be constantly working, without any time gaps between frames because the CPU is easily able to feed commands to the GPU before the GPU is done processing them. Instead, it doesn't seem like the CPU is 3 frames ahead. It seems like the CPU is constantly waiting for the GPU to be finished before it starts working on a new frame. So it makes me wonder if my triple buffering code is possibly broken? Here is my code for moving to the next frame:
void gpu_interface::next_frame()
{
UINT64 current_frame_fence_value = get_frame_resource()->fence_value;
UINT64 next_frame_fence_value = current_frame_fence_value + 1;
check_hr(swapchain->Present(0, 0));
check_hr(graphics_cmd_queue->Signal(fence.Get(), current_frame_fence_value));
{
// CPU and GPU frame-to-frame event.
PIXEndEvent(graphics_cmd_queue.Get());
PIXBeginEvent(graphics_cmd_queue.Get(), 0, "fence value: %d", next_frame_fence_value);
}
// Check if the next frame is ready to be rendered.
// The GPU must have reached at least up to the fence value of the frame we're about to render.
if (fence->GetCompletedValue() < current_frame_fence_value)
{
PIXBeginEvent(0, "CPU Waiting for GPU to reach fence value: %d", current_frame_fence_value);
// Wait for the next frame resource to be ready
fence->SetEventOnCompletion(current_frame_fence_value, fence_event);
WaitForSingleObject(fence_event, INFINITE);
PIXEndEvent();
}
// Next frame is ready to be rendered
// Update the frame_index. GetCurrentBackBufferIndex() gets incremented after swapchain->Present() calls.
frame_index = swapchain->GetCurrentBackBufferIndex();
frames[frame_index].fence_value = next_frame_fence_value;
}
Here's the whole timing capture: https://1drv.ms/u/s!AiGFMy6hVmtNgaky52n7QDrQ6o7V1A?e=MFc4xW
EDIT: Fixed answer
void gpu_interface::next_frame()
{
check_hr(swapchain->Present(0, 0));
UINT64 current_frame_fence_value = get_frame_resource()->fence_value;
UINT64 next_frame_fence_value = current_frame_fence_value + 1;
check_hr(graphics_cmd_queue->Signal(fence.Get(), current_frame_fence_value));
//// Update the frame_index. GetCurrentBackBufferIndex() gets incremented after swapchain->Present() calls.
frame_index = swapchain->GetCurrentBackBufferIndex();
// The GPU must have reached at least up to the fence value of the frame we're about to render.
size_t minimum_fence = get_frame_resource()->fence_value;
size_t completed = fence->GetCompletedValue();
if (completed < minimum_fence)
{
PIXBeginEvent(0, "CPU Waiting for GPU to reach fence value: %d", minimum_fence);
// Wait for the next frame resource to be ready
fence->SetEventOnCompletion(minimum_fence, fence_event);
WaitForSingleObject(fence_event, INFINITE);
PIXEndEvent();
}
frames[frame_index].fence_value = next_frame_fence_value;
{
// CPU and GPU frame-to-frame event.
PIXEndEvent(graphics_cmd_queue.Get());
PIXBeginEvent(graphics_cmd_queue.Get(), 0, "fence value: %d", next_frame_fence_value);
}
}
Timing capture of the correct code: https://1drv.ms/u/s!AiGFMy6hVmtNgakzGizTiA_s-FwPqA?e=qIHHTw
You signal the queue with current_frame_fence_value and right after you check
if (fence->GetCompletedValue() < current_frame_fence_value)
if the fence completed that value. You need to check the fence value for the next frame to see if you can continue and that is fence_values[frame_index] where frame_index is updated. It would go something like this:
void gpu_interface::next_frame()
{
check_hr(swapchain->Present(0, 0));
UINT64 current_frame_fence_value = get_frame_resource()->fence_value;
check_hr(graphics_cmd_queue->Signal(fence.Get(), current_frame_fence_value));
UINT64 next_frame_fence_value = current_frame_fence_value + 1;
frame_index = swapchain->GetCurrentBackBufferIndex();
// The GPU must have reached at least up to the fence value of the frame we're about to render.
//current_frame_fence_value is not the fence value of the frame you are about the render, it is fence_values[frame_index]
//note that frame_index is updated before this call
if (fence->GetCompletedValue() < fence_values[frame_index])
{
// Wait for the next frame resource to be ready
fence->SetEventOnCompletion(fence_values[frame_index], fence_event);
WaitForSingleObject(fence_event, INFINITE);
}
frames[frame_index].fence_value = next_frame_fence_value;
}
Try writing down fence values for the first few frames to see how that works.
My first app synthesised music audio from a sine look-up table using methods deprecated since iOS 6. I have just revised it to address warnings about AudioSessionhelped by this blog and the Apple guidelines on AVFoundationFramework. Audio Session warnings have now been addressed and the app produces audio as it did before. It currently runs under iOS 9.
However the app occasionally crashes for no apparent reason. I checked out this SO post but it seems to deal with accessing rather than generating raw audio data, so maybe it is not dealing with a timing issue. I suspect there is a buffering problem but I need to understand what this might be before I change or fine tune anything in the code.
I have a deadline to make the revised app available to users so I'd be most grateful to hear from someone who has dealt a similar issue.
Here is the issue. The app goes into debug on the simulator reporting:
com.apple.coreaudio.AQClient (8):EXC_BAD_ACCESS (code=1, address=0xffffffff10626000)
In the Debug Navigator, Thread 8 (com.apple.coreaudio.AQClient (8)), it reports:
0 -[Synth fillBuffer:frames:]
1 -[PlayView audioBufferPlayer:fillBuffer:format:]
2 playCallback
This line of code in fillBuffer is highlighted
float sineValue = (1.0f - b)*sine[a] + b*sine[c];
... and so is this line of code in audioBufferPlayer
int packetsWritten = [synth fillBuffer:buffer->mAudioData frames:packetsPerBuffer];
... and playCallBack
[player.delegate audioBufferPlayer:player fillBuffer:inBuffer format:player.audioFormat];
Here is the code for audioBufferPlayer (delegate, essentially the same as in the demo referred to above).
- (void)audioBufferPlayer:(AudioBufferPlayer*)audioBufferPlayer fillBuffer:(AudioQueueBufferRef)buffer format:(AudioStreamBasicDescription)audioFormat
{
[synthLock lock];
int packetsPerBuffer = buffer->mAudioDataBytesCapacity / audioFormat.mBytesPerPacket;
int packetsWritten = [synth fillBuffer:buffer->mAudioData frames:packetsPerBuffer];
buffer->mAudioDataByteSize = packetsWritten * audioFormat.mBytesPerPacket;
[synthLock unlock];
}
... (initialised in myViewController)
- (id)init
{
if ((self = [super init])) {
// The audio buffer is managed (filled up etc.) within its own thread (Audio Queue thread)
// Since we are also responding to changes from the GUI, we need a lock so both threads
// do not attempt to change the same value independently.
synthLock = [[NSLock alloc] init];
// Synth and the AudioBufferPlayer must use the same sample rate.
float sampleRate = 44100.0f;
// Initialise synth to fill the audio buffer with audio samples.
synth = [[Synth alloc] initWithSampleRate:sampleRate];
// Initialise note buttons
buttons = [[NSMutableArray alloc] init];
// Initialise the audio buffer.
player = [[AudioBufferPlayer alloc] initWithSampleRate:sampleRate channels:1 bitsPerChannel:16 packetsPerBuffer:1024];
player.delegate = self;
player.gain = 0.9f;
[[AVAudioSession sharedInstance] setActive:YES error:nil];
}
return self;
} // initialisation
... and for playCallback
static void playCallback( void* inUserData, AudioQueueRef inAudioQueue, AudioQueueBufferRef inBuffer)
{
AudioBufferPlayer* player = (AudioBufferPlayer*) inUserData;
if (player.playing){
[player.delegate audioBufferPlayer:player fillBuffer:inBuffer format:player.audioFormat];
AudioQueueEnqueueBuffer(inAudioQueue, inBuffer, 0, NULL);
}
}
... and here is the code for fillBuffer where audio is synthesised
- (int)fillBuffer:(void*)buffer frames:(int)frames
{
SInt16* p = (SInt16*)buffer;
// Loop through the frames (or "block size"), then consider each sample for each tone.
for (int f = 0; f < frames; ++f)
{
float m = 0.0f; // the mixed value for this frame
for (int n = 0; n < MAX_TONE_EVENTS; ++n)
{
if (tones[n].state == STATE_INACTIVE) // only active tones
continue;
// recalculate a 30sec envelope and place in a look-up table
// Longer notes need to interpolate through the envelope
int a = (int)tones[n].envStep; // integer part (like a floored float)
float b = tones[n].envStep - a; // decimal part (like doing a modulo)
// c allows us to calculate if we need to wrap around
int c = a + 1; // (like a ceiling of integer part)
if (c >= envLength) c = a; // don't wrap around
/////////////// LOOK UP ENVELOPE TABLE /////////////////
// uses table look-up with interpolation for both level and pitch envelopes
// 'b' is a value interpolated between 2 successive samples 'a' and 'c')
// first, read values for the level envelope
float envValue = (1.0f - b)*tones[n].levelEnvelope[a] + b*tones[n].levelEnvelope[c];
// then the pitch envelope
float pitchFactorValue = (1.0f - b)*tones[n].pitchEnvelope[a] + b*tones[n].pitchEnvelope[c];
// Advance envelope pointer one step
tones[n].envStep += tones[n].envDelta;
// Turn note off at the end of the envelope.
if (((int)tones[n].envStep) >= envLength){
tones[n].state = STATE_INACTIVE;
continue;
}
// Precalculated Sine look-up table
a = (int)tones[n].phase; // integer part
b = tones[n].phase - a; // decimal part
c = a + 1;
if (c >= sineLength) c -= sineLength; // wrap around
///////////////// LOOK UP OF SINE TABLE ///////////////////
float sineValue = (1.0f - b)*sine[a] + b*sine[c];
// Wrap round when we get to the end of the sine look-up table.
tones[n].phase += (tones[n].frequency * pitchFactorValue); // calculate frequency for each point in the pitch envelope
if (((int)tones[n].phase) >= sineLength)
tones[n].phase -= sineLength;
////////////////// RAMP NOTE OFF IF IT HAS BEEN UNPRESSED
if (tones[n].state == STATE_UNPRESSED) {
tones[n].gain -= 0.0001;
if ( tones[n].gain <= 0 ) {
tones[n].state = STATE_INACTIVE;
}
}
//////////////// FINAL SAMPLE VALUE ///////////////////
float s = sineValue * envValue * gain * tones[n].gain;
// Clip the signal, if needed.
if (s > 1.0f) s = 1.0f;
else if (s < -1.0f) s = -1.0f;
// Add the sample to the out-going signal
m += s;
}
// Write the sample mix to the buffer as a 16-bit word.
p[f] = (SInt16)(m * 0x7FFF);
}
return frames;
}
I'm not sure whether it is a red herring but I came across NaN in several debug registers. It appears to happen while calculating phase increment for sine lookup in fillBuffer (see above). That calculation is done for up to a dozen partials every sample at a sampling rate of 44.1 kHz and worked in iOS 4 on an iPhone 4. I'm running on simulator of iOS 9. The only changes I made are described in this post!
My NaN problem turned out to have nothing directly to do with Core Audio. It was caused by an edge condition introduced by changes in another area of my code. The real problem was a division by zero attempted while calculating the duration of the sound envelope in realtime.
However, in trying to identify the cause of that problem, I am confident my pre-iOS 7 Audio Session has been replaced by a working setup based on AVFoundation. Thanks goes to the source of my initial code Matthijs Hollemans and also to Mario Diana whose blog explained the changes needed.
At first, the sound levels on my iPhone were significantly less than the sound levels on the Simulator, a problem addressed here by foundry. I found it necessary to include these improvements by replacing Mario's
- (BOOL)setUpAudioSession
with foundry's
- (void)configureAVAudioSession
Hopefully this might help someone else.
I would like to use the EZAudio framework to do realtime microphone signal FFT processing, along with some other processing in order to determine the peak frequency.
The problem is, the EZmicrophone class only appears to work on 512 samples, however, my signal requires an FFT of 8192 or even 16384 samples. There doesnt appear to be a way to change the buffer size in EZMicrophone, but I've read posts that recommend creating an array of my target size and appending the microphone buffer to it, then when it's full, do the FFT.
When I do this though, I get large chunks of memory with no data, or discontinuities between the segments of copied memory. I think it may have something to do with the timing or order in which the microphone delegate is being called or memory being overwritten in different threads...I'm grasping at straws here. Am I correct in assuming that this code is being executed everytime the microphone buffer is full of a new 512 samples?
Can anyone suggest what I may be doing wrong? I've been stuck on this for a long time.
Here is the post I've been using as a reference:
EZAudio: How do you separate the buffersize from the FFT window size(desire higher frequency bin resolution).
// Global variables which are bad but I'm just trying to make things work
float tempBuf[512];
float fftBuf[8192];
int samplesRemaining = 8192;
int samplestoCopy = 512;
int FFTLEN = 8192;
int fftBufIndex = 0;
#pragma mark - EZMicrophoneDelegate
-(void) microphone:(EZMicrophone *)microphone
hasAudioReceived:(float **)buffer
withBufferSize:(UInt32)bufferSize
withNumberOfChannels:(UInt32)numberOfChannels {
// Copy the microphone buffer so it wont be changed
memcpy(tempBuf, buffer[0], bufferSize);
dispatch_async(dispatch_get_main_queue(),^{
// Setup the FFT if it's not already setup
if( !_isFFTSetup ){
[self createFFTWithBufferSize:FFTLEN withAudioData:fftBuf];
_isFFTSetup = YES;
}
int samplesRemaining = FFTLEN;
memcpy(fftBuf+fftBufIndex, tempBuf, samplestoCopy*sizeof(float));
fftBufIndex += samplestoCopy;
samplesRemaining -= samplestoCopy;
if (fftBufIndex == FFTLEN)
{
fftBufIndex = 0;
samplesRemaining = FFTLEN;
[self updateFFTWithBufferSize:FFTLEN withAudioData:fftBuf];
}
});
}
You likely have threading issues because you are trying to do work in some blocks that takes much much longer than the time between audio callbacks. Your code is being called repeatedly before prior calls can say that they are done (with the FFT setup or clearing the FFT buffer).
Try doing the FFT setup outside the callback before starting the recording, only copy to a circular buffer or FIFO inside the callback, and do the FFT in code async to the callback (not locked in the same block as the circular buffer copy).
I'm working with a small Pololu 3pi robot. It has 32KB of flash, 2KB of SRAM and 1KB of EEPROM. I ran the following code on the robot:
#include <pololu/3pi.h>
int main(){
int nums[1000];
nums[0] = 50;
nums[999] = 100;
clear();
print_long(nums[0]); // prints 50
lcd_goto_xy(0, 1);
print_long(nums[999]); // prints 100
while(1);
}
My expectation was that it would crash because it's run out of RAM
to store the entire nums array. But not only did it not crash, it also printed the numbers correctly as they were allocated.
How come? Isn't this using 4000 bytes of memory?