Is possible to compare a voice with already recorded voice in the phone.Based on the comparison we can rate like Good, Very Good , Excellent etc. Most closed sound get high rating.
Anybody know is it possible in Android?
Help is highly appreciable.
For a general audio processing library I can recommend marsyas. Unfortunately the official home page is currently down.
Marsyas even provides a sample android application. After getting a proper signal analysis framework, you need to analyse your signal. For example, the AimC implementation for marsyas can be used to compare voice.
I recommend installing marsyas on your computer and fiddle with the python example scripts.
For your voice analysis, you could use a network like this:
vqNetwork = ["Series/vqlizer", [
"AimPZFC/aimpzfc",
"AimHCL/aimhcl",
"AimLocalMax/aimlocalmax",
"AimSAI/aimsai",
"AimBoxes/aimBoxes",
"AimVQ/vq",
"Gain/g",
]
This network takes your audio data and transforms it as it would be processed by a human ear. After that it uses vector quantization to reduce the many possible vectors to very specific codebooks with 200 entries. You can then translate the output of the network to readable characters (utf8 for example), which you then can compare using something like string edit distances (e.g. Levenshtein distance).
Another possibility is to use MFCC (Mel Frequency Cepstral Coefficients) for speech recognition which marsyas supports as well and use something, for example Dynamic Time Warping, to compare the outputs. This document describes the process pretty well.
Using 'Musicg' library you can compare two voice (.wav format) files.
use Wave object to load the wave file to instantiate in pgm.
here using FingerPrintSimilarity
function you pass pre recorded wav files to get the output.
But you should know that "musicg" library deals only with .wav format files, so if you have a an .mp3 file for example you need to convert it to a wave file first.
android gradle dependency:
implementation group: 'com.github.fracpete', name: 'musicg', version: '1.4.2.2'
for more:
https://github.com/loisaidasam/musicg
sample code:
private void compareTempFile(String str) {
Wave w1 = new Wave(Environment.getExternalStorageDirectory().getAbsolutePath()+"/sample1.wav");
Wave w2 = new Wave(Environment.getExternalStorageDirectory().getAbsolutePath()+"/sample2.wav");
println("Wave 1 = "+w1.getWaveHeader());
println("Wave 2 = "+w2.getWaveHeader());
FingerprintSimilarity fpsc1 = w2.getFingerprintSimilarity(w1);
float scorec = fpsc1.getScore();
float simc= fpsc1.getSimilarity();
tvSim.setText(" Similarity = "+simc+"\nScore = "+scorec);
println("Score = "+scorec);
println("Similarity = "+simc);
}
Related
I am working on Android ExoPlayer as mentioned in this Article- https://betterprogramming.pub/android-exoplayer-play-videos-in-your-app-like-youtube-486853913397
But I am unable to control the video quality (360p, 480p, etc). How to do that? I need the complete code.
For ABR streams, ExoPlayer will automatically switch to the best bit rate based on its assessment of current network conditions etc - e.g. if it is playing a high bit rate for a particular piece of content and determines that the network is busy and its buffer is not keeping up, it will switch to a lower bit rate for that content. More info here:
https://exoplayer.dev/track-selection.html
If you mean that you would like to be able to control the bit rate manually, then the track selector functionality will allow you do that.
You can see more info here (linked from the ExoPLayer GitHub): https://medium.com/google-exoplayer/exoplayer-2-x-track-selection-2b62ff712cc9
The default interface looks like:
You can select the quality version by creating your own ABR algorithm, e.g., yourOwnABR(), then call it in the function updateSelectedTrack() of file AdaptiveTrackSelection.java as follows.
public void updateSelectedTrack(
long playbackPositionUs,
long bufferedDurationUs,
long availableDurationUs,
List<? extends MediaChunk> queue,
MediaChunkIterator[] mediaChunkIterators) {
...
int newSelectedIndex = yourOwnABR();
...
I have spent much time trying to find out where is my mistakes while Im flashing the PIC16F688. The Pic has successfully flashed using PicKit2. Im using the Pic to convert analog pressure sensor to digital output and sending the data via Bluetooth, but the Bluetooth is not receiving stable numbers of data. The data is consist of 4 character decimal number that is between 0 and 1023.
The problem is that the Bluetooth can't wait at specific number and keep reading it, instead, it is reading the 4 digits in random.
I think my mistake is within the configuration of internal oscillator.
I'm attaching my code, the code is written to configure the flexiforce sensor circuit that outputs analog voltage up to 5v, and then the pic duty is to convert it to digital as I mentioned above.
it might be my wiring is not correct, please If you could help out with this one
and what configuration "at edit project" do I need to choose for Mikro PRO software?
I used "Bluetooth terminal" app to see my data asynchronous from Bluetooth.
Thank you.
char *temp = "0000";
unsigned int adc_value;
char uart_rd; int i;
void main()
{
OSCCON = 0x77;
ANSEL = 0b00000100;
CMCON0 = 0X07;
TRISA = 0b00001100;
UART1_Init(9600);
Delay_ms(100);
while (1)
{
adc_value = ADC_Read(2);
temp[0] = adc_value/1000+48;
temp[1] = (adc_value/100)%10+48;
temp[2] = (adc_value/10)%10+48;
temp[3] = adc_value%10+48;
for (i=0;i<4; i++)
UART1_Write(temp[i]);
UART1_Write(13);
Delay_ms(1000);
}
}
You can use itoa function to convert ADC integer value to characters for sending over UART. If there is error in calculation then you wont get appropriate value. Below code snippet for your reference :
while (1)
{
adc_value = ADC_Read(2);
itoa(adc_value, temp, 10);
for (i=0;i<4; i++)
UART1_Write(temp[i]);
UART1_Write(13);
Delay_ms(1000);
}
Please check Baud Rate you have configured at both ends is same or not. If baudrate mismatches then you will get Random value at Bluetooth Terminal where you are reading values.
What i would suggest, if you have a logic analyser, hook it up. If you don't recalculate your oscillator speed with the datasheet. It could just be that the internal oscillator is not accurate enough. What also works, is to write a function in assembly that waits a known time (by copy-pasting a lot of NOPs and using this to blink a led. Then start a stopwatch and count, say, 100 blinks. This is what i used to do before i had a logic analyser. (They are quite cheep on ebay).
I am transcoding videos based on the example given by Google (https://android.googlesource.com/platform/cts/+/master/tests/tests/media/src/android/media/cts/ExtractDecodeEditEncodeMuxTest.java)
Basically, transocding of MP4 files works, but on some phones I get some weird results. If for example I transcode a video with audio on an HTC One, the code won't give any errors but the file cannot play afterward on the phone. If I have a 10 seconds video it jumps to almost the last second and you only here some crackling noise. If you play the video with VLC the audio track is completely muted.
I did not alter the code in terms of encoding/decoding and the same code gives correct results on a Nexus 5 or MotoX for example.
Anybody having an idea why it might fail on that specific device?
Best regard and thank you,
Florian
I made it work in Android 4.4.2 devices by following changes:
Set AAC profile to AACObjectLC instead of AACObjectHE
private static final int OUTPUT_AUDIO_AAC_PROFILE = MediaCodecInfo.CodecProfileLevel.AACObjectLC;
During creation of output audio format, use sample rate and channel count of input format instead of fixed values
MediaFormat outputAudioFormat = MediaFormat.createAudioFormat(OUTPUT_AUDIO_MIME_TYPE,
inputFormat.getInteger(MediaFormat.KEY_SAMPLE_RATE),
inputFormat.getInteger(MediaFormat.KEY_CHANNEL_COUNT));
Put a check just before audio muxing audio track to control presentation timestamps. (To avoid timestampUs X < lastTimestampUs X for Audio track error)
if (audioPresentationTimeUsLast == 0) { // Defined in the begining of method
audioPresentationTimeUsLast = audioEncoderOutputBufferInfo.presentationTimeUs;
} else {
if (audioPresentationTimeUsLast > audioEncoderOutputBufferInfo.presentationTimeUs) {
audioEncoderOutputBufferInfo.presentationTimeUs = audioPresentationTimeUsLast + 1;
}
audioPresentationTimeUsLast = audioEncoderOutputBufferInfo.presentationTimeUs;
}
// Write data
if (audioEncoderOutputBufferInfo.size != 0) {
muxer.writeSampleData(outputAudioTrack, encoderOutputBuffer, audioEncoderOutputBufferInfo);
}
Hope this helps...
If original CTS tests fail you need to go to device vendors and ask for fixes
I am doing a video compression project for Android and I am thinking of implementing it by designing a new video codec (by scratch , I have designed the algorithm) . I have already read the basics of video compression , related relevant algorithms and codec basics . I have also found that FFmpeg may serve as a quite good solution on Android.
Now my questions come:
How to write a new video codec as in FFmpeg? I am still a beginner at writing codecs , but
how do I start ? I have a rough idea that that you have to write at least a demuxer first and then the specific encoder and decoder etc . (Asking for references here please.)
Since my codec deosn't simply adjust video properties like fps , resolution , bit-rate etc.
Is reading the MediaCodec API and MediaPlayer API in official Android SDK enough for writing new codecs ? (Because last time I saw it had only support for MPEG-4 SP , H.263 and H.264 . I was unable to find if you could directly write your own classes and functions).
Thanks .
You can use ffmpeg as a tool or the ffmpeg set of libraries (libavcodec, libaviformat, …) on Android. You can add or change ffmpeg codecs in a cross- platform manner, because this project puts a strong emphasis on platform independence. You can use the MediaCodec API instead. But there is no way to extend the MediaCodec API (update it is possible to extend MediaCodec, it is documented at http://source.android.com/devices/media.html#codecs ) and no easy way to let ffmpeg use this API.
if you are a newb and "just want to do it in SW", than just do it in SW. I am assuming your algorithm does not need to be real-time, and compress video data on the fly, or you would need to use a HW codec.
This is from Android MediaCodec Reference
MediaCodec codec = MediaCodec.createDecoderByType(type);
codec.configure(format, ...);
codec.start();
ByteBuffer[] inputBuffers = codec.getInputBuffers();
ByteBuffer[] outputBuffers = codec.getOutputBuffers();
for (;;) {
int inputBufferIndex = codec.dequeueInputBuffer(timeoutUs);
if (inputBufferIndex >= 0) {
// fill inputBuffers[inputBufferIndex] with valid data
...
codec.queueInputBuffer(inputBufferIndex, ...);
}
int outputBufferIndex = codec.dequeueOutputBuffer(timeoutUs);
if (outputBufferIndex >= 0) {
// outputBuffer is ready to be processed or rendered.
...
codec.releaseOutputBuffer(outputBufferIndex, ...);
} else if (outputBufferIndex == MediaCodec.INFO_OUTPUT_BUFFERS_CHANGED) {
outputBuffers = codec.getOutputBuffers();
} else if (outputBufferIndex == MediaCodec.INFO_OUTPUT_FORMAT_CHANGED) {
// Subsequent data will conform to new format.
MediaFormat format = codec.getOutputFormat();
...
}
}
codec.stop();
codec.release();
codec = null;
On the line that reads "// outputBuffer is ready to be processed or rendered" apply your codec.
That is your first frame will be outputBuffers[0] to outputBuffers[outputBufferIndex]. Save off outputBufferIndex, i.e. outputBufferIndex_old = outputBufferIndex; then your next frame will be outputbuffers[outputBufferIndex_old] to outputbuffers[outputBufferIndex]. But this is a circular buffer, so in the for loop ... ahhhhh
something like this:
//init
int old = 0;
int len = codec.BufferInfo().size,buff_len=outputBuffers.size;
Byte[] processBuffer = new Byte[len];
... // outputBuffer ready
for (int i=old; i<old+len; i++){
processBuffer[i-old] = outputBuffers[i%buff_len];
}
old = outputBufferIndex;
Here is a good example. You may want to look into MediaMetadataRetriever to get information about the input video. height and width ect. bytesize per pixel, if you want your encoder to be robust to different types of video. Anyway, that should get you started.
I strongly recommend Matlab(or GNU Octave) for prototyping a video codec. It will save you a ton of time. Meaning you should make sure your intended codec algorithm works before trying to implement it on a near impossible system to debug like Android.
Hope this helps.
If someone stumbles across this old question the answer is:
Write your Program.
Where you want the "Codec" to go simply add a 'null Codec' (copy Input to Output).
Test that your Program still works and that you can read the (so-called) encoded File.
Add your Codec where the 'null Codec' was (call a Function to avoid big edits to a working File).
Re-Test your Program to ensure it still works and read the Output to make sure it is correct.
That is all. ;)
Things to consider:
A "Video Player" can drop Frames, a "Video Recorder" had better NOT
drop Frames.
A 'Software Codec' (no Hardware assist) will be slow,
run it on a different Core, if available.
A Hardware Codec (called from Software) will be necessary unless you are just making a
Demo.
Split your Program into pieces that can run separately so it can be threaded and those Threads can be assigned to different Cores. You will need to detect the number of Cores and assess their speed so you can do some of the partitioning dynamically at Runtime.
Use of the NDK and Assembly Language Programming will be necessary to get enough speed to compress a decent sized Video at a wanted frame rate (IE: you do not want your finished Program to only support 320x176 # 5 FPS Videos). The Compressor MUST run faster than it's Input arrives.
Designing your own Codec to beat an existing Codec (x265) will take you years (without help).
If your a Wiz at Java, C, and ARM Assembly (and a Software Engineer) it will take more than a couple of months of work; so commit or quit. Try to find some Open Source as a base to start from.
I have some audio data (raw AAC) inside a byte array for playback. During playback, I need to get its volume/amplitude to draw (something like an audio wave when playing).
What I'm thinking now is to get the volume/amplitude of the current audio every 200 milliseconds and use that for drawing (using a canvas), but I'm not sure how to do that.
.
.
.
.
** 2011/07/13 add following **
Sorry just been delayed on other project until now.
What I tried is run the following codes in a thread, and playing my AAC audio.
a loop
{
// int v=audio.getStreamVolume(AudioManager.MODE_NORMAL);
// int v=audio.getStreamVolume(AudioManager.STREAM_MUSIC);
int v=audio.getStreamVolume(AudioManager.STREAM_DTMF);
// Tried 3 settings above
Log.i(HiCardConstants.TAG, "Volume - "+v);
try{Thread.sleep(200);}
catch(InterruptedException ie){}
}
But only get a fixed value, not dynamic volume...
And I also found a class named Visualizer, but unfortunately, my target platform is Android 2.2 ... :-(
Any suggestions are welcome :-)
After days and nights, I found that an Android app project called ringdroid
can solve my problem.
It helps me to get an audio gain value array, so that I can use to to draw my sound wave.
BTW, as my experience, some .AMR or .MP3 can't be parsed correctly, due to too low bitrate...