I am trying to transmit audio using CAN FD and the data format it sends it is in uint8_t, when the sample I take from the microphone with the ADC is an uint32_t. Will I have any problem trying to insert this uint8_t data into a wav after being transmitted with CAN FD? I have tried to create .wav with the data but I only hear noise.
This is and image of a .wav file I create with a simple 64 byte CAN FD message.
.wav file
I know only introducing 64 bytes is a really small amount for a .wav but I dont know how to concatenate more data into a buffer in order to get bigger files.
Thiis the function I use to create my .wav
void write_little_endian(unsigned int word, int num_bytes, FILE *wav_file)
{
uint16_t buf;
while(num_bytes>0)
{ buf = word & 0xff;
fwrite(&buf, 1,1, wav_file);
num_bytes--;
word >>= 8;
}
}
void write_wav(char * filename, unsigned long num_samples, short int * data, int s_rate)
{
FILE* wav_file;
unsigned int sample_rate;
unsigned int num_channels;
unsigned int bytes_per_sample;
unsigned int byte_rate;
unsigned long i; /* counter for samples */
num_channels = 1; /* monoaural */
bytes_per_sample = 1;
if (s_rate<=0) sample_rate = 16000;
else sample_rate = (unsigned int) s_rate;
byte_rate = sample_rate*num_channels*bytes_per_sample;
wav_file = fopen(filename, "wb");
assert(wav_file); /* make sure it opened */
/* write RIFF header */
fwrite("RIFF", 1, 4, wav_file);
write_little_endian(36 + bytes_per_sample* num_samples*num_channels, 4, wav_file);
fwrite("WAVE", 1, 4, wav_file);
/* write fmt subchunk */
fwrite("fmt ", 1, 4, wav_file);
write_little_endian(16, 4, wav_file); /* SubChunk1Size is 16 */
write_little_endian(1, 2, wav_file); /* PCM is format 1 */
write_little_endian(num_channels, 2, wav_file);
write_little_endian(sample_rate, 4, wav_file);
write_little_endian(byte_rate, 4, wav_file);
write_little_endian(num_channels*bytes_per_sample, 2, wav_file); /* block align */
write_little_endian(8*bytes_per_sample, 2, wav_file); /* bits/sample */
/* write data subchunk */
fwrite("data", 1, 4, wav_file);
write_little_endian(bytes_per_sample* num_samples*num_channels, 4, wav_file);
for (i=0; i< num_samples; i++)
{ write_little_endian((unsigned int)(data[i]),bytes_per_sample, wav_file);
}
fclose(wav_file);
}
I send this command to call it and I receive what I will post on the next image:
write_wav("test.wav", 44100, (short int *)buffer, 44100);
.wav with a bigger buffer of 44100 samples of uint16_t
But when I play this file, the audio is just beeps and noise, not my voice
:(
Update = After some tries, I think the problem resides in the data I send throught the CAN FD. I have watched the data inside another .wav and the format doesnt look similar to mines. I dont know how to send uint32_t data that I acquire with the ADC through a uint8_t format CAN FD data field. I have tried casting the data but I dont know how to prepare it for the .wav Can anyone help me with that? Should I send it with uint8_t and then back to uin32_t before inserting it into a .wav?
It depends on what sample rate you need. At 5 Mbps baudrate, 11 bit identifiers, 64 byte data, the effective payload will be transferred at roughly 4 Mbps (this can be calculated fairly exactly). The MCU will need a bit of execution time to process it, but it's probably negligible. That is, the real time spec by far exceeds at what sample rates the human ear stop hearing any differences at. Your problems aren't related to picking CAN FD as hardware/datalink layer. You could likely do this with classic CAN just fine too.
Obviously don't transfer files but raw data. It only needs to be turned into a file before playing.
Related
I am trying to send measured i2s analogue signal (e.g. from mic) to the sink device via Bluetooth instead of the default noise.
Currently I am trying to change the bt_app_a2d_data_cb()
static int32_t bt_app_a2d_data_cb(uint8_t *data, int32_t i2s_read_len)
{
if (i2s_read_len < 0 || data == NULL) {
return 0;
}
char* i2s_read_buff = (char*) calloc(i2s_read_len, sizeof(char));
bytes_read = 0;
i2s_adc_enable(I2S_NUM_0);
while(bytes_read == 0)
{
i2s_read(I2S_NUM_0, i2s_read_buff, i2s_read_len,&bytes_read, portMAX_DELAY);
}
i2s_adc_disable(I2S_NUM_0);
// taking care of the watchdog//
TIMERG0.wdt_wprotect=TIMG_WDT_WKEY_VALUE;
TIMERG0.wdt_feed=1;
TIMERG0.wdt_wprotect=0;
uint32_t j = 0;
uint16_t dac_value = 0;
// change 16bit input signal to 8bit
for (int i = 0; i < i2s_read_len; i += 2) {
dac_value = ((((uint16_t) (i2s_read_buff[i + 1] & 0xf) << 8) | ((i2s_read_buff[i + 0]))));
data[j] = (uint8_t) dac_value * 256 / 4096;
j++;
}
// testing for loop
//uint8_t da = 0;
//for (int i = 0; i < i2s_read_len; i++) {
// data[i] = (uint8_t) (i2s_read_buff[i] >> 8);// & 0xff;
// da++;
// if(da>254) da=0;
//}
free(i2s_read_buff);
i2s_read_buff = NULL;
return i2s_read_len;
}
I can hear the sawtooth sound from the sink device.
Any ideas what to do?
your data can be an array of some float digits representing analog signals or analog signal variations, for example, a 32khz sound signal contains 320000 float numbers to define captures sound for every second. if your data have been expected to transmit in offline mode you can prepare your outcoming data in the form of a buffer plus a terminator sign then send buffer by Bluetooth module of sender device which is connected to the proper microcontroller. for the receiving device, if you got terminator character like "\r" you can process incoming buffer e.g. for my case, I had to send a string array of numbers but I often received at most one or two unknown characters and to avoid it I reject it while fulfill receiving container.
how to trim unknown first characters of string in code vision
if you want it in online mode i.e. your data must be transmitted and played concurrently. you must consider delays and reasonable time to process for all microcontrollers and devices like Bluetooth, EEprom iCs and...
I'm also working on a project "a2dp source esp32".
I'm playing a wav-file from spiffs.
If the wav-file is 44100, 16-bit, stereo then you can directly write a stream of bytes from the file to the array data[ ].
When I tried to write less data than in the len-variable and return less (for example 88), I got an error, now I'm trying to figure out how to reduce this buffer because of big latency (len=512).
Also, the data in the array data[ ] is stored as stereo.
Example: read data from file to data[ ]-array:
size_t read;
read = fread((void*) data, 1, len, fwave);//fwave is a file
if(read<len){//If get EOF, go to begin of the file
fseek(fwave , 0x2C , SEEK_SET);//skip wav-header 44bytesт
read = fread((void*) (&(data[read])), 1, len-read, fwave);//read up
}
If file mono, I convert it to stereo like this (I read half and then double data):
int32_t lenHalf=len/2;
read = fread((void*) data, 1, lenHalf, fwave);
if(read<lenHalf){
fseek(fwave , 0x2C , SEEK_SET);//skip wav-header 44bytesт
read = fread((void*) (&(data[read])), 1, lenHalf-read, fwave);//read up
}
//copy to the second channel
uint16_t *data16=(uint16_t*)data;
for (int i = lenHalf/2-1; i >= 0; i--) {
data16[(i << 1)] = data16[i];
data16[(i << 1) + 1] = data16[i];
}
I think you have got sawtooth sound because:
your data is mono?
in your "return i2s_read_len;" i2s_read_len less than len
you // change 16bit input signal to 8bit, in the array data[ ] data as 16-bit: 2ByteLeft-2ByteRight-2ByteLeft-2ByteRight-...
I'm not sure, it's a guess.
I am using Lame's mpglib to decode mp3 to PCM in Android NDK for playing. But when I called hip_decode(), it returen 0 meaning that "need more data before we can complete the decode". I had no idea how to solve it. Can someone helps me? Here is my code:
void CBufferWrapper::ConvertMp3toPCM (AAssetManager* mgr, const char *filename){
Print ("ConvertMp3toPCM:file:%s", filename);
AAsset* asset = AAssetManager_open (mgr, filename, AASSET_MODE_UNKNOWN);
// the asset might not be found
assert (asset != NULL);
// open asset as file descriptor
off_t start, length;
int fd = AAsset_openFileDescriptor (asset, &start, &length);
assert (0 <= fd);
long size = AAsset_getLength (asset);
char* buffer = (char*)malloc (sizeof(char)*size);
memset (buffer, 0, size*sizeof(char));
AAsset_read (asset, buffer, size);
AAsset_close (asset);
hip_t ht = hip_decode_init ();
int count = hip_decode (ht, (unsigned char*)buffer, size, pcm_l, pcm_r);
free (buffer);
Print ("ConvertMp3toPCM: length:%ld,pcmcount=%d",length, count);
}
I used MACRO "HAVE_MPGLIB" to compile Lame in NDK. So I think it should work for decoding literally.
Yesterday I had the same problem. Is the same problem but using lame_enc.dll. I did not know how to resolve this 0 returned, this is the reason to this post.
Create a buffer to put mp3 data: unsigned char mp3Data[4096]
Create two buffers for pcm data, but bigger than mp3 one:
unsigned short[4096 * 100];
Open mp3 file and initialize hip.
Now, enter in a do while loop until read bytes are 0 (the end of file).
Inside the loop read 4096 bytes into mp3Data and call hip_decode with
hip_decode(ht, mp3Data, bytesRead, lpcm, rpcm);
You are right, it returns 0. It is asking you for more data.
You need to repeat the reading of 4096 bytes and the call to hip_decode until it returns a valid samples number.
Here is the important part of my program:
int total = 0;
int hecho = 0;
int leido = 0;
int lon = 0;
int x;
do
{
total = fread(mp3b, 1, MAXIMO, fich);
leido += total;
x = hip_decode(hgf, mp3b, total, izquierda, derecha);
if(x > 0)
{
int tamanio;
int y;
tamanio = 1.45 * x + 9200;
unsigned char * bu = (unsigned char *) malloc(tamanio);
y = lame_encode_buffer(lamglofla, izquierda, derecha, x, bu, tamanio);
fwrite(bu, 1, y, fichs);
free(bu);
}
}while(total > 0);
My program decodes a mp3 file and encodes the output into another mp3 file.
I expect that this could be useful.
I have a very simple task to do, but somehow I am still stuck.
I have one BIG data file ("File_initial.dat"), which should be read by all nodes on the cluster (using MPI), each node will perform some manipulation on part of this BIG file (File_size / number_of_nodes) and finally each node will write its result to one shared BIG file ("File_final.dat"). The number of elements of files remain the same.
By googling I understood, that it is much better to write data file as a binary file (I have only decimal numbers in this file) and not as *.txt" file. Since no human will read this file, but only computers.
I tried to implement myself (but using formatted in/output and NOT binary file) this, but I get incorrect behavior.
My code so far follows:
#include <fstream>
#define NNN 30
int main(int argc, char **argv)
{
ifstream fin;
// setting MPI environment
int rank, nprocs;
MPI_File file;
MPI_Init(&argc, &argv);
MPI_Comm_size(MPI_COMM_WORLD, &nprocs);
MPI_Comm_rank(MPI_COMM_WORLD, &rank);
// reading the initial file
fin.open("initial.txt");
for (int i=0;i<NNN;i++)
{
fin >> res[i];
cout << res[i] << endl; // to see, what I have in the file
}
fin.close();
// starting position in the "res" array as a function of "rank" of process
int Pstart = (NNN / nprocs) * rank ;
// specifying Offset for writing to file
MPI_Offset offset = sizeof(double)*rank;
MPI_File file;
MPI_Status status;
// opening one shared file
MPI_File_open(MPI_COMM_WORLD, "final.txt", MPI_MODE_CREATE|MPI_MODE_WRONLY,
MPI_INFO_NULL, &file);
// setting local for each node array
double * localArray;
localArray = new double [NNN/nprocs];
// Performing some basic manipulation (squaring each element of array)
for (int i=0;i<(NNN / nprocs);i++)
{
localArray[i] = res[Pstart+i]*res[Pstart+i];
}
// Writing the result of each local array to the shared final file:
MPI_File_seek(file, offset, MPI_SEEK_SET);
MPI_File_write(file, localArray, sizeof(double), MPI_DOUBLE, &status);
MPI_File_close(&file);
MPI_Finalize();
return 0;
}
I understand, that I do something wrong, while trying to write double as a text file.
How one should change the code in order to be able to save
as .txt file (format output)
as .dat file (binary file)
Your binary file output is almost right; but your calculations for your offset within the file and the amount of data to write is incorrect. You want your offset to be
MPI_Offset offset = sizeof(double)*Pstart;
not
MPI_Offset offset = sizeof(double)*rank;
otherwise you'll have each rank overwriting each others data as (say) rank 3 out of nprocs=5 starts writing at double number 3 in the file, not (30/5)*3 = 18.
Also, you want each rank to write NNN/nprocs doubles, not sizeof(double) doubles, meaning you want
MPI_File_write(file, localArray, NNN/nprocs, MPI_DOUBLE, &status);
How to write as a text file is a much bigger issue; you have to convert the data into string internally and then output those strings, making sure you know how many characters each line requires by careful formatting. That is described in this answer on this site.
I am decoding aac to pcm with ffmpeg with avcodec_decode_audio3. However it decodes into AV_SAMPLE_FMT_FLTP sample format (PCM 32bit Float Planar) and i need AV_SAMPLE_FMT_S16 (PCM 16 bit signed - S16LE).
I know that ffmpeg can do this easily with -sample_fmt. I want to do the same with the code but i still couldn't figure it out.
audio_resample did not work for: it fails with error message: .... conversion failed.
EDIT 9th April 2013: Worked out how to use libswresample to do this... much faster!
At some point in the last 2-3 years FFmpeg's AAC decoder's output format changed from AV_SAMPLE_FMT_S16 to AV_SAMPLE_FMT_FLTP. This means that each audio channel has it's own buffer, and each sample value is a 32-bit floating point value scaled from -1.0 to +1.0.
Whereas with AV_SAMPLE_FMT_S16 the data is in a single buffer, with the samples interleaved, and each sample is a signed integer from -32767 to +32767.
And if you really need your audio as AV_SAMPLE_FMT_S16, then you have to do the conversion yourself. I figured out two ways to do it:
1. Use libswresample (recommended)
#include "libswresample/swresample.h"
...
SwrContext *swr;
...
// Set up SWR context once you've got codec information
swr = swr_alloc();
av_opt_set_int(swr, "in_channel_layout", audioCodec->channel_layout, 0);
av_opt_set_int(swr, "out_channel_layout", audioCodec->channel_layout, 0);
av_opt_set_int(swr, "in_sample_rate", audioCodec->sample_rate, 0);
av_opt_set_int(swr, "out_sample_rate", audioCodec->sample_rate, 0);
av_opt_set_sample_fmt(swr, "in_sample_fmt", AV_SAMPLE_FMT_FLTP, 0);
av_opt_set_sample_fmt(swr, "out_sample_fmt", AV_SAMPLE_FMT_S16, 0);
swr_init(swr);
...
// In your decoder loop, after decoding an audio frame:
AVFrame *audioFrame = ...;
int16_t* outputBuffer = ...;
swr_convert(&outputBuffer, audioFrame->nb_samples, audioFrame->extended_data, audioFrame->nb_samples);
And that's all you have to do!
2. Do it by hand in C (original answer, not recommended)
So in your decode loop, when you've got an audio packet you decode it like this:
AVCodecContext *audioCodec; // init'd elsewhere
AVFrame *audioFrame; // init'd elsewhere
AVPacket packet; // init'd elsewhere
int16_t* outputBuffer; // init'd elsewhere
int out_size = 0;
...
int len = avcodec_decode_audio4(audioCodec, audioFrame, &out_size, &packet);
And then, if you've got a full frame of audio, you can convert it fairly easily:
// Convert from AV_SAMPLE_FMT_FLTP to AV_SAMPLE_FMT_S16
int in_samples = audioFrame->nb_samples;
int in_linesize = audioFrame->linesize[0];
int i=0;
float* inputChannel0 = (float*)audioFrame->extended_data[0];
// Mono
if (audioFrame->channels==1) {
for (i=0 ; i<in_samples ; i++) {
float sample = *inputChannel0++;
if (sample<-1.0f) sample=-1.0f; else if (sample>1.0f) sample=1.0f;
outputBuffer[i] = (int16_t) (sample * 32767.0f);
}
}
// Stereo
else {
float* inputChannel1 = (float*)audioFrame->extended_data[1];
for (i=0 ; i<in_samples ; i++) {
outputBuffer[i*2] = (int16_t) ((*inputChannel0++) * 32767.0f);
outputBuffer[i*2+1] = (int16_t) ((*inputChannel1++) * 32767.0f);
}
}
// outputBuffer now contains 16-bit PCM!
I've left a couple of things out for clarity... the clamping in the mono path should ideally be duplicated in the stereo path. And the code can be easily optimized.
I found 2 resample function from FFMPEG. The performance maybe better.
avresample_convert()
http://libav.org/doxygen/master/group__lavr.html
swr_convert() http://spirton.com/svn/MPlayer-SB/ffmpeg/libswresample/swresample_test.c
Thanks Reuben for a solution to this. I did find that some of the sample values were slightly off when compared with a straight ffmpeg -i file.wav. It seems that in the conversion, they use a round() on the value.
To do the conversion, I did what you did with a bid of modification to work for any amount of channels:
if (audioCodecContext->sample_fmt == AV_SAMPLE_FMT_FLTP)
{
int nb_samples = decoded_frame->nb_samples;
int channels = decoded_frame->channels;
int outputBufferLen = nb_samples & channels * 2;
short* outputBuffer = new short[outputBufferLen/2];
for (int i = 0; i < nb_samples; i++)
{
for (int c = 0; c < channels; c++)
{
float* extended_data = (float*)decoded_frame->extended_data[c];
float sample = extended_data[i];
if (sample < -1.0f) sample = -1.0f;
else if (sample > 1.0f) sample = 1.0f;
outputBuffer[i * channels + c] = (short)round(sample * 32767.0f);
}
}
// Do what you want with the data etc.
}
I went from ffmpeg 0.11.1 -> 1.1.3 and found the change of sample format annoying. I looked at setting the request_sample_fmt to AV_SAMPLE_FMT_S16 but it seems the aac decoder doesn't support anything other than AV_SAMPLE_FMT_FLTP anyway.
Does the WaveOut API has some internal limitation of the size for the current piece of buffer played ? I mean if I provide a very small buffer does it affects somehow the sound played to the speakers. I am experiencing very strange noise when I am generating and playing the sinus wave with small buffer. Something like a peak, or "BUMP".
The complete Story:
I made a program that can generate Sinus sound signal in real time.
The variable parameters are Frequency and Volume. The project requirement was to have a maximum latency of 50 ms. So the program must be able to produce Sinus signals with manually adjustable frequency of audio signal in real time.
I used Windows WaveOut API, C# and P/invoke to access the API.
Everything works fine when the sound buffer is 1000 ms large. If I minimize the buffer to 50 ms as per latency requirement then for certain frequencies I am experiencing at the end of every buffer, a noise or "BUMP". I do not understand if the sound generated is malformed ( I checked and is not) or something happens with the Audio chip, or some delay in initializing and playing.
When I save the produced audio to .wav file everything is perfect.
This means the must be some bug in my code or the audio subsystem has a limitation to the buffer chunks sent to it.
For those who doesn't know WaveOut must be initialized at first time and then must be prepared with audio headers for each buffer that are containing the number of bytes that needs to be played and the pointer to a memory that contains the audio that needs to be player.
UPDATE
Noise happens with the following combinations 44100 SamplingRate, 16 Bits, 2 channels, 50 ms buffer and generated Sinus audio signal of 201Hz, 202Hz, 203Hz, 204Hz, 205Hz ... 219Hz,
220Hz, 240 Hz, is ok
Why is this difference of 20, I do not know.
There are a few things to keep in mind when you need to output audio smoothly:
waveOutXxxx API is a legacy/compatibility layer on top of lower level API and as such it has greater overhead and is not recommended when you are to reach minimal latency. Note that this is unlikely to be your primary problem, but this is a piece of general knowledge helpful for understanding
because Windows is not real time OS and its audio subsystem is not realtime either you don't have control over random latency involved between you queue audio data for output and the data is really played back, the key is to keep certain level of buffer fullness which protects you from playback underflows and delivers smooth playback
with waveOutXxxx you are no limited to having single buffer, you can allocate multiple reusable buffers and recycle them
All in all, waveOutXxxx, DirectSound, DirectShow APIs work well with latencies 50 ms and up. With WASAPI exclusive mode streams you can get 5 ms latencies and even lower.
EDIT: I seem to have said too early about 20 ms latencies. To compensate for this, here is a simple tool LowLatencyWaveOutPlay (Win32, x64) to estimate the latency you can achieve. With sufficient buffering playback is smooth, otherwise you hear stuttering.
My understanding is that buffers might be returned late and the optimal design in terms of smallest latency lies along the line of having more smaller buffers so that you are given them back as early as possible. For example, 10 buffers 3 ms/buffer rather than 3 buffers 10 ms/buffer.
D:\>LowLatencyWaveOutPlay.exe 48000 10 3
Format: 48000 Hz, 1 channels, 16 bits per sample
Buffer Count: 10
Buffer Length: 3 ms (288 bytes)
Signal Frequency: 1000 Hz
^C
So I came here because I wanted to find the basic latency of waveoutwrite() as well. I got around 25-26ms of latency before I got to the smooth sine tone.
This is for:
AMD Phenom(tm) 9850 Quad-Core Processor 2.51 GHz
4.00 GB ram
64-bit operating system, x64-based processor
Windows 10 Enterprise N
The code follows. It is a modfied version of Petzold's sine wave program, refactored to run on the command line. I also changed the polling of buffers to use of a callback on buffer complete with the idea that this would make the program more efficient, but it didn't make a difference.
It also has a setup for elapsed timing, which I used to probe various timings for operations on the buffers. Using those I get:
Sine wave output program
Channels: 2
Sample rate: 44100
Bytes per second: 176400
Block align: 4
Bits per sample: 16
Time per buffer: 0.025850
Total time prepare header: 87.5000000000 usec
Total time to fill: 327.9000000000 usec
Total time for waveOutWrite: 90.8000000000 usec
Program:
/*******************************************************************************
WaveOut example program
Based on C. Petzold's sine wave example, outputs a sine wave via the waveOut
API in Win32.
*******************************************************************************/
#include <stdio.h>
#include <windows.h>
#include <math.h>
#include <limits.h>
#include <unistd.h>
#define SAMPLE_RATE 44100
#define FREQ_INIT 440
#define OUT_BUFFER_SIZE 570*4
#define PI 3.14159
#define CHANNELS 2
#define BITS 16
#define MAXTIM 1000000000
double fAngle;
LARGE_INTEGER perffreq;
PWAVEHDR pWaveHdr1, pWaveHdr2;
int iFreq = FREQ_INIT;
VOID FillBuffer (short* pBuffer, int iFreq)
{
int i;
int c;
for (i = 0 ; i < OUT_BUFFER_SIZE ; i += CHANNELS) {
for (c = 0; c < CHANNELS; c++)
pBuffer[i+c] = (short)(SHRT_MAX*sin (fAngle));
fAngle += 2*PI*iFreq/SAMPLE_RATE;
if (fAngle > 2 * PI) fAngle -= 2*PI;
}
}
double elapsed(LARGE_INTEGER t)
{
LARGE_INTEGER rt;
long tt;
QueryPerformanceCounter(&rt);
tt = rt.QuadPart-t.QuadPart;
return (tt*(1.0/(double)perffreq.QuadPart));
}
void CALLBACK waveOutProc(HWAVEOUT hwo, UINT uMsg, DWORD_PTR dwInstance, DWORD_PTR dwParam1, DWORD_PTR dwParam2)
{
if (uMsg == WOM_DONE) {
if (pWaveHdr1->dwFlags & WHDR_DONE) {
FillBuffer((short*)pWaveHdr1->lpData, iFreq);
waveOutWrite(hwo, pWaveHdr1, sizeof(WAVEHDR));
}
if (pWaveHdr2->dwFlags & WHDR_DONE) {
FillBuffer((short*)pWaveHdr2->lpData, iFreq);
waveOutWrite(hwo, pWaveHdr2, sizeof(WAVEHDR));
}
}
}
int main()
{
HWAVEOUT hWaveOut ;
short* pBuffer1;
short* pBuffer2;
short* pBuffer3;
WAVEFORMATEX waveformat;
UINT wReturn;
int bytes;
long t;
LARGE_INTEGER rt;
double timprep;
double filtim;
double waveouttim;
printf("Sine wave output program\n");
fAngle = 0; /* start sine angle */
QueryPerformanceFrequency(&perffreq);
pWaveHdr1 = malloc (sizeof (WAVEHDR));
pWaveHdr2 = malloc (sizeof (WAVEHDR));
pBuffer1 = malloc (OUT_BUFFER_SIZE*sizeof(short));
pBuffer2 = malloc (OUT_BUFFER_SIZE*sizeof(short));
pBuffer3 = malloc (OUT_BUFFER_SIZE*sizeof(short));
if (!pWaveHdr1 || !pWaveHdr2 || !pBuffer1 || !pBuffer2) {
if (!pWaveHdr1) free (pWaveHdr1) ;
if (!pWaveHdr2) free (pWaveHdr2) ;
if (!pBuffer1) free (pBuffer1) ;
if (!pBuffer2) free (pBuffer2) ;
fprintf(stderr, "*** Error: No memory\n");
exit(1);
}
// Load prime parameters to format
waveformat.wFormatTag = WAVE_FORMAT_PCM;
waveformat.nChannels = CHANNELS;
waveformat.nSamplesPerSec = SAMPLE_RATE;
waveformat.wBitsPerSample = BITS;
waveformat.cbSize = 0;
// Calculate other parameters
bytes = waveformat.wBitsPerSample/8; /* find bytes per sample */
if (waveformat.wBitsPerSample&8) bytes++; /* round up */
bytes *= waveformat.nChannels; /* find total channels size */
waveformat.nBlockAlign = bytes; /* set block align */
/* find average bytes/sec */
waveformat.nAvgBytesPerSec = bytes*waveformat.nSamplesPerSec;
printf("Channels: %d\n", waveformat.nChannels);
printf("Sample rate: %d\n", waveformat.nSamplesPerSec);
printf("Bytes per second: %d\n", waveformat.nAvgBytesPerSec);
printf("Block align: %d\n", waveformat.nBlockAlign);
printf("Bits per sample: %d\n", waveformat.wBitsPerSample);
printf("Time per buffer: %f\n",
OUT_BUFFER_SIZE*sizeof(short)/(double)waveformat.nAvgBytesPerSec);
if (waveOutOpen (&hWaveOut, WAVE_MAPPER, &waveformat, (DWORD_PTR)waveOutProc, 0, CALLBACK_FUNCTION)
!= MMSYSERR_NOERROR) {
free (pWaveHdr1) ;
free (pWaveHdr2) ;
free (pBuffer1) ;
free (pBuffer2) ;
hWaveOut = NULL ;
fprintf(stderr, "*** Error: No memory\n");
exit(1);
}
// Set up headers and prepare them
pWaveHdr1->lpData = (LPSTR)pBuffer1;
pWaveHdr1->dwBufferLength = OUT_BUFFER_SIZE*sizeof(short);
pWaveHdr1->dwBytesRecorded = 0;
pWaveHdr1->dwUser = 0;
pWaveHdr1->dwFlags = WHDR_DONE;
pWaveHdr1->dwLoops = 1;
pWaveHdr1->lpNext = NULL;
pWaveHdr1->reserved = 0;
QueryPerformanceCounter(&rt);
waveOutPrepareHeader(hWaveOut, pWaveHdr1, sizeof (WAVEHDR));
timprep = elapsed(rt);
pWaveHdr2->lpData = (LPSTR)pBuffer2;
pWaveHdr2->dwBufferLength = OUT_BUFFER_SIZE*sizeof(short);
pWaveHdr2->dwBytesRecorded = 0;
pWaveHdr2->dwUser = 0;
pWaveHdr2->dwFlags = WHDR_DONE;
pWaveHdr2->dwLoops = 1;
pWaveHdr2->lpNext = NULL;
pWaveHdr2->reserved = 0;
waveOutPrepareHeader(hWaveOut, pWaveHdr2, sizeof (WAVEHDR));
// Send two buffers to waveform output device
QueryPerformanceCounter(&rt);
FillBuffer (pBuffer1, iFreq);
filtim = elapsed(rt);
QueryPerformanceCounter(&rt);
waveOutWrite (hWaveOut, pWaveHdr1, sizeof (WAVEHDR));
waveouttim = elapsed(rt);
FillBuffer (pBuffer2, iFreq);
waveOutWrite (hWaveOut, pWaveHdr2, sizeof (WAVEHDR));
// Run waveform loop
sleep(10);
printf("Total time prepare header: %.10f usec\n", timprep*1000000);
printf("Total time to fill: %.10f usec\n", filtim*1000000);
printf("Total time for waveOutWrite: %.10f usec\n", waveouttim*1000000);
waveOutUnprepareHeader(hWaveOut, pWaveHdr1, sizeof (WAVEHDR));
waveOutUnprepareHeader(hWaveOut, pWaveHdr2, sizeof (WAVEHDR));
// Close waveform file
free (pWaveHdr1) ;
free (pWaveHdr2) ;
free (pBuffer1) ;
free (pBuffer2) ;
}