android AudioTrack playback short array (16bit) - audio

I have an application that playback audio. It takes encoded audio data over RTP and decode it to 16bit array. The decoded 16bit array is converted to 8 bit array (byte array) as this is required for some other functionality.
Even though audio playback is working it is breaking continuously and very hard to recognise audio output. If I listen carefully I can tell it is playing the correct audio.
I suspect this is due to the fact I convert 16 bit data stream into a byte array and use the write(byte[], int, int, AudioTrack.WRITE_NON_BLOCKING) of AudioTrack class for audio playback.
Therefore I converted the byte array back to a short array and used write(short[], int, int, AudioTrack.WRITE_NON_BLOCKING) method to see if it could resolve the problem.
However now there is no audio sound at all. In the debug output I can see the short array has data.
What could be the reason?
Here is the AUdioTrak initialization
sampleRate =AudioTrack.getNativeOutputSampleRate(AudioManager.STREAM_MUSIC);
minimumBufferSize = AudioTrack.getMinBufferSize(sampleRate, AudioFormat.CHANNEL_OUT_STEREO, AudioFormat.ENCODING_PCM_16BIT);
audioTrack = new AudioTrack(AudioManager.STREAM_MUSIC, sampleRate,
AudioFormat.CHANNEL_OUT_STEREO,
AudioFormat.ENCODING_PCM_16BIT,
minimumBufferSize,
AudioTrack.MODE_STREAM);
Here is the code converts short array to byte array
for (int i=0;i<internalBuffer.length;i++){
bufferIndex = i*2;
buffer[bufferIndex] = shortToByte(internalBuffer[i])[0];
buffer[bufferIndex+1] = shortToByte(internalBuffer[i])[1];
}
Here is the method that converts byte array to short array.
public short[] getShortAudioBuffer(byte[] b){
short audioBuffer[] = null;
int index = 0;
int audioSize = 0;
ByteBuffer byteBuffer = ByteBuffer.allocate(2);
if ((b ==null) && (b.length<2)){
return null;
}else{
audioSize = (b.length - (b.length%2));
audioBuffer = new short[audioSize/2];
}
if ((audioSize/2) < 2)
return null;
byteBuffer.order(ByteOrder.LITTLE_ENDIAN);
for(int i=0;i<audioSize/2;i++){
index = i*2;
byteBuffer.put(b[index]);
byteBuffer.put(b[index+1]);
audioBuffer[i] = byteBuffer.getShort(0);
byteBuffer.clear();
System.out.print(Integer.toHexString(audioBuffer[i]) + " ");
}
System.out.println();
return audioBuffer;
}
Audio is decoded using opus library and the configuration is as follows;
opus_decoder_ctl(dec,OPUS_SET_APPLICATION(OPUS_APPLICATION_AUDIO));
opus_decoder_ctl(dec,OPUS_SET_SIGNAL(OPUS_SIGNAL_MUSIC));
opus_decoder_ctl(dec,OPUS_SET_FORCE_CHANNELS(OPUS_AUTO));
opus_decoder_ctl(dec,OPUS_SET_MAX_BANDWIDTH(OPUS_BANDWIDTH_FULLBAND));
opus_decoder_ctl(dec,OPUS_SET_PACKET_LOSS_PERC(0));
opus_decoder_ctl(dec,OPUS_SET_COMPLEXITY(10)); // highest complexity
opus_decoder_ctl(dec,OPUS_SET_LSB_DEPTH(16)); // 16bit = two byte samples
opus_decoder_ctl(dec,OPUS_SET_DTX(0)); // default - not using discontinuous transmission
opus_decoder_ctl(dec,OPUS_SET_VBR(1)); // use variable bit rate
opus_decoder_ctl(dec,OPUS_SET_VBR_CONSTRAINT(0)); // unconstrained
opus_decoder_ctl(dec,OPUS_SET_INBAND_FEC(0)); // no forward error correction

Let's assume you have a short[] array which contains the 16-bit one channel data to be played.
Then each sample is a value between -32768 and 32767 which represents the signal amplitude at the exact moment. And 0 value represents a middle point (no signal). This array can be passed to the audio track with ENCODING_PCM_16BIT format encoding.
But things are going weird when playing ENCODING_PCM_8BIT is used (See AudioFormat)
In this case each sample encoded by one byte. But each byte is unsigned. That means, it's value is between 0 and 255, while 128 represents the middle point.
Java has no unsigned byte format. Byte format is signed. I.e. values -128...-1 will represent actual values of 128...255. So you have to be careful when converting to the byte array, otherwise it will be a noise with barely recognizable source sound.
short[] input16 = ... // the source 16-bit audio data;
byte[] output8 = new byte[input16.length];
for (int i = 0 ; i < input16.length ; i++) {
// To convert 16 bit signed sample to 8 bit unsigned
// We add 128 (for rounding), then shift it right 8 positions
// Then add 128 to be in range 0..255
int sample = ((input16[i] + 128) >> 8) + 128;
if (sample > 255) sample = 255; // strip out overload
output8[i] = (byte)(sample); // cast to signed byte type
}
To perform backward conversion all should be the same: each single sample to be converted to exactly one sample of the output signal
byte[] input8 = // source 8-bit unsigned audio data;
short[] output16 = new short[input8.length];
for (int i = 0 ; i < input8.length ; i++) {
// to convert signed byte back to unsigned value just use bitwise AND with 0xFF
// then we need subtract 128 offset
// Then, just scale up the value by 256 to fit 16-bit range
output16[i] = (short)(((input8[i] & 0xFF) - 128) * 256);
}

The issue of not being able to convert data from byte array to short array was resolved when used bitwise operators instead of using ByteArray. It could be due not setting the correct parameters in ByteArray or it is not suitable for such conversion.
Nevertheless implementing conversion using bitwise operators resolved the problem. Since the original question has been resolved by this approach, please consider this as the final answer.
I will raise a separate topic for playback issue.
Thank you for all your support.

Related

sending audio via bluetooth a2dp source esp32

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.

Dump subtitle from AVSubtitle in the file

In FFMPEG sofftware, AVPicture is used to store image data using data pointer and linesizes.It means all subtitles are stored in the form of picture inside ffmpeg. Now I have DVB subtitle and I want to dump picture of subtitles stored in AVPicture in a buffer. I know these images subtitles can be dump using for, fopen and sprintf. But do not know how to dump Subtitle.I have to dump subtitles in .ppm file format.
Can anyone help me to dump picture of subtitles in buffer from AVSubtitle .
This process looks complex but actually very simple.
AVSubtitle is generic format, supports text and bitmap modes. Dvbsub format afaik bitmap only and the bitmap format can be differ like 16color or 256color mode as called CLUT_DEPTH.
I believe (in current ffmpeg) the bitmaps stored in AVSubtitleRect structure, which is member of AVSubtitle.
I assume you have a valid AVSubtitle packet(s) and if I understand correctly you can do these and it should work:
1) Check pkt->rect[0]->type. The pkt here is a valid AVSubtitle packet. It must be type of SUBTITLE_BITMAP.
2) If so, bitmap with and height can be read from pkt->rects[0]->w and pkt->rects[0]->h.
3) Bitmap data itself in will be pkt->rects[0]->data[0].
4) CLUT_DEPTH can be read from pkt->rects[0]->nb_colors.
5) And CLUT itself (color table) will be in pkt->rects[0]->data[1].
With these data, you can construct a valid .bmp file that can be viewable on windows or linux desktop, but I left this part to you.
PPM Info
First check this info about PPM format:
https://www.cs.swarthmore.edu/~soni/cs35/f13/Labs/extras/01/ppm_info.html
What I understand is PPM format uses RGB values (24bit/3bytes). It looks like to me all you have to do is construct a header according to data obtained from AVSubtitle packet above. And write a conversion function for dvbsub's indexed color buffer to RGB. I'm pretty sure somewhere there are some ready to use codes out there but I'll explain anyway.
In the picture frame data Dvbsub uses is liner and every pixel is 1 byte (even in 16color mode). This byte value is actually index value that correspond RGB (?) values stored in Color Look-Up Table (CLUT), in 16 color mode there are 16 index each 4 bytes, first 3 are R, G, B values and 4th one is alpha (transparency values, if PPM doesn't support this, ignore it).
I'm not sure if decoded subtitle still has encoded YUV values. I remember it should be plain RGBA format.
encode_dvb_subtitles function on ffmpeg shows how this encoding done. If you need it.
https://github.com/FFmpeg/FFmpeg/blob/a0ac49e38ee1d1011c394d7be67d0f08b2281526/libavcodec/dvbsub.c
Hope that helps.
As this is where I ended up when searching for answers to how to create a thumbnail of an AVSubtitle, here is what I ended up using in my test application. The code is optimized for readability only. I got some help from this question which had some sample code.
Using avcodec_decode_subtitle2() I get a AVSubtitle structure. This contains a number of rectangles. First I iterate over the rectangles to find the max of x + w and y + h to determine the width and height of the target frame.
The color table in data[1] is RGBA, so I allocate an AVFrame called frame in AV_PIX_FMT_RGBA format and shuffle the pixels over to it:
struct [[gnu::packed]] rgbaPixel {
uint8_t r;
uint8_t g;
uint8_t b;
uint8_t a;
};
// Copy the pixel buffers
for (unsigned int i = 0; i < sub.num_rects; ++ i) {
AVSubtitleRect* rect = sub.rects[i];
for (int y = 0; y < rect->h; ++ y) {
int dest_y = y + rect->y;
// data[0] holds index data
uint8_t *in_linedata = rect->data[0] + y * rect->linesize[0];
// In AVFrame, data[0] holds the pixel buffer directly
uint8_t *out_linedata = frame->data[0] + dest_y * frame->linesize[0];
rgbaPixel *out_pixels = reinterpret_cast<rgbaPixel*>(out_linedata);
for (int x = 0; x < rect->w; ++ x) {
// data[1] contains the color map
// compare libavcodec/dvbsubenc.c
uint8_t colidx = in_linedata[x];
uint32_t color = reinterpret_cast<uint32_t*>(rect->data[1])[colidx];
// Now store the pixel in the target buffer
out_pixels[x + rect->x] = rgbaPixel{
.r = static_cast<uint8_t>((color >> 16) & 0xff),
.g = static_cast<uint8_t>((color >> 8) & 0xff),
.b = static_cast<uint8_t>((color >> 0) & 0xff),
.a = static_cast<uint8_t>((color >> 24) & 0xff),
};
}
}
}
I did manage to push that AVFrame through an image decoder to output it as a bitmap image, and it looked OK. I did get green areas where the alpha channel is, but that might be an artifact of the settings in the JPEG encoder I used.

Converting 24 bit USB audio stream into 32 bit stream

I'm trying to convert a 24 bit usb audio stream into a 32 bit stream so my microcontroller's peripherals can play happily with the stream (it can only handle 16 or 32 bit data like most mcus...).
The following code is what I got from the mcu's company... didn't work as expected and I ended up getting really distorted audio.
// Function takes usb stream and processes the data for our peripherals
// #data - usb stream data
// #byte_count - size of stream
void process_usb_stream(uint8_t *data, uint16_t byte_count) {
// Etc code that gets buffers ready to read the stream...
// Conversion here!
int32_t *buffer;
int sample_count = 0;
for (int i = 0; i < byte_count; i += 3) {
buffer[sample_count++] = data[i] | data[i+1] << 8 | data[i+2] << 16;
}
// Send buffer to peripherals for them to use...
}
Any help with converting the data from a 24 bit stream to 32 bit stream would be super awesome! This area of work is very hard for me :(
data[...] is a uint8_t. You need to cast that before shifting, because data[...]<<8 and data[...]<<16 are undefined. They'll either be 0 or unchanged, neither of which is what you want.
Also, you need to shift by another 8 bits to get the full range and put the sign bit in the right place.
Also, you're treating the data as if it were in little-endian format. Make sure it is. I'll assume that's correct, so something like this works:
int32_t *buffer;
int sample_count = 0;
for (int i = 0; i+3 <= byte_count; ) {
int32_t v = ((int32_t)data[i++])<<8;
v |= ((int32_t)data[i++])<<16;
v |= ((int32_t)data[i++])<<24;
buffer[sample_count++] = v;
}
Finally, note that this assumes that byte_count is divisible by 3 -- make sure that's true!
this is DSP stuff if, also post this question on http://dsp.stackexchange.com
In DSP the process of changing the bit depth is called scaling
16 bit resolution has 65536 values
24 bit resolution has 16777216
possible values
32 bit has 4294967296 values so the factor is 256
According to https://electronics.stackexchange.com/questions/229268/what-is-name-of-process-used-to-change-sample-bit-depth/229271
reduction from 24 bit to 16 bit is called scaling down and is done by dividing each value by 256.
This can be done by bitwise shifting every bit by 8
y = x >> 8. When scaling down this way the LSB is lost
Scaling up to 32 bit is more complicated and there are several approaches how to do this. It may work by multiplying each bit of the value with a value between 2⁰ and 2⁸.
Push the 24 bit value in a 32 bit register and then left-shifting each bit by a value between 2⁰ and 2⁸:
data32[31] = data32[23] << 8;
data32[22] = data32[14] << 8;
...
data32[0] = data32[0];
and interpolate the bits you do not get with this (linear interpolation)
Maybe there are much better scaling up algortihms ask on http://dsp.stackexchange.com
See also http://blog.bjornroche.com/2013/05/the-abcs-of-pcm-uncompressed-digital.html for the scaling up problem...

How to send an int over uint8_t data?

I'm using the RadioHead Packet Radio library from airspayce.com. In the example (nrf24_reliable_datagram_client & server) they let two nodes communicate with each other by sending strings back and forth. Now I want to send an int instead of a string there, and do something with this data. This is what they do in the example:
Define the buf byte.
uint8_t buf[RH_NRF24_MAX_MESSAGE_LEN];
This function receives the data:
manager.recvfromAckTimeout(buf, &len, 500, &from)
Print the buf variable.
Serial.print((char*)buf);
So far so good.Now I want to do something like:
int value = (char*)buf;
Or:
char value[10] = { (char*)buf };
But then I get:
invalid conversion from 'char*' to 'int' (or to 'char'...)
Next to that, on the other side where I'm sending the data, I have:
uint8_t data[] = { analogRead(A0) };
When I'm printing this data on the receiver side, using the code from the first question, I get weird characters. So I thought, let's try:
Serial.print((char*)buf, DEC); // or BYTE
But then I get:
call of overloaded 'print(char*, int)' is ambiguous
What am I doing wrong? Thanks in advance!
You can't just assign an array to an integer and hope that it merges the elements together for you - for example, how does it know how to merge them?
For converting a uint16_t to a uint8_t[2] array you would want to do something like this:
uint16_t analog = analogRead(A0); //read in as int.
uint8_t data[2] = {analog, (analog >> 8)}; // extract as {lower byte, upper byte)
Serial.write(data,2); //write the two bytes to the serial port, lower byte first.
You could do it in other ways like using a union of a uint16_t with an array of two uint8_t's, but the above way is more portable. You could also do it by type casting the pointer to an int, however if one end uses big endian and the other uses little endian, that won't work unless you flip the data around in the array as you are receiving it.
For the receiver end, you would have:
uint8_t data[2];
...
... //whatever you do to receive the bytes that were sent over serial.
...
//Now assuming that data[] contains the received bytes where:
//data[0] was the first in (lower byte) and data[1] was the second in (upper byte)
uint16_t merged = (data[1] << 8) | data[0]; //merge them back together
Hopefully that helps.
Also, the 'overloaded prototype' is saying that no function exists which takes that particular set of input variables. From the print class header you will find there is however this prototype:
write(const uint8_t *buffer, size_t size);
which does what you want - print a specified number of uint8_t's from an array.

Converting an int or String to a char array on Arduino

I am getting an int value from one of the analog pins on my Arduino. How do I concatenate this to a String and then convert the String to a char[]?
It was suggested that I try char msg[] = myString.getChars();, but I am receiving a message that getChars does not exist.
To convert and append an integer, use operator += (or member function concat):
String stringOne = "A long integer: ";
stringOne += 123456789;
To get the string as type char[], use toCharArray():
char charBuf[50];
stringOne.toCharArray(charBuf, 50)
In the example, there is only space for 49 characters (presuming it is terminated by null). You may want to make the size dynamic.
Overhead
The cost of bringing in String (it is not included if not used anywhere in the sketch), is approximately 1212 bytes of program memory (flash) and 48 bytes RAM.
This was measured using Arduino IDE version 1.8.10 (2019-09-13) for an Arduino Leonardo sketch.
Risk
There must be sufficient free RAM available. Otherwise, the result may be lockup/freeze of the application or other strange behaviour (UB).
Just as a reference, below is an example of how to convert between String and char[] with a dynamic length -
// Define
String str = "This is my string";
// Length (with one extra character for the null terminator)
int str_len = str.length() + 1;
// Prepare the character array (the buffer)
char char_array[str_len];
// Copy it over
str.toCharArray(char_array, str_len);
Yes, this is painfully obtuse for something as simple as a type conversion, but somehow it's the easiest way.
You can convert it to char* if you don't need a modifiable string by using:
(char*) yourString.c_str();
This would be very useful when you want to publish a String variable via MQTT in arduino.
None of that stuff worked. Here's a much simpler way .. the label str is the pointer to what IS an array...
String str = String(yourNumber, DEC); // Obviously .. get your int or byte into the string
str = str + '\r' + '\n'; // Add the required carriage return, optional line feed
byte str_len = str.length();
// Get the length of the whole lot .. C will kindly
// place a null at the end of the string which makes
// it by default an array[].
// The [0] element is the highest digit... so we
// have a separate place counter for the array...
byte arrayPointer = 0;
while (str_len)
{
// I was outputting the digits to the TX buffer
if ((UCSR0A & (1<<UDRE0))) // Is the TX buffer empty?
{
UDR0 = str[arrayPointer];
--str_len;
++arrayPointer;
}
}
With all the answers here, I'm surprised no one has brought up using itoa already built in.
It inserts the string representation of the integer into the given pointer.
int a = 4625;
char cStr[5]; // number of digits + 1 for null terminator
itoa(a, cStr, 10); // int value, pointer to string, base number
Or if you're unsure of the length of the string:
int b = 80085;
int len = String(b).length();
char cStr[len + 1]; // String.length() does not include the null terminator
itoa(b, cStr, 10); // or you could use String(b).toCharArray(cStr, len);

Resources