Is there any way to record audio in high quality?
And how can I read information that user is saying something? In Audio Recording application you can see such indicator (I don't know the right name for it).
At the moment, a big reason for poor audio quality recording on Android is the codec used by the MediaRecorder class (the AMR-NB codec). However, you can get access to uncompressed audio via the AudioRecord class, and record that into a file directly.
The Rehearsal Assistant app does this to save uncompressed audio into a WAV file - take a look at the RehearsalAudioRecord class source code.
The RehearsalAudioRecord class also provides a getMaxAmplitude method, which you can use to detect the maximum audio level since the last time you called the method (MediaRecorder also provides this method).
For recording and monitoring: You can use the sound recorder activity.
Here's a snippet of code:
Intent recordIntent = new Intent(
MediaStore.Audio.Media.RECORD_SOUND_ACTION);
startActivityForResult(recordIntent, REQUEST_CODE_RECORD);
For a perfect working example of how to record audio which includes an input monitor, download the open source Ringdroid project: https://github.com/google/ringdroid
Look at the screenshots and you'll see the monitor.
For making the audio higher quality, you'd need a better mic. The built in mic can only capture so much (which is not that good). Again, look at the ringdroid project, glean some info from there. At that point you could implement some normalization and amplification routines to improve the sound.
I give you a simple answer.
for samplerate, about the quality, 48000 is almost the same as 16000.
for bitrate, about the quality, 96Kbps is much better than 16Kbps.
you can try stereo(channelCount = 2), but make little change.
So, for android phones, just set the audio bit rate bigger, you will get the better quality.
Related
I have an app calling using WebRTC. But during a call, I need to record microphone. WebRTC has an object WebRTCAudioRecord to record audio but the audio file is so large (PCM_16bit). I want to record but to a smaller size.
I've tried MediaRecorder but it doesn't work because WebRTC is recorded and MediaRecorder does not have permission to record while calling.
Has anyone done this, or have any idea that could help me?
Webrtc is considered as comparatively much better pre-processing tool for Audio and Video.
Webrtc native development includes fully optimized native C and C++ classes, In order to maintain wonderful Speech Quality and Intelligibility of audio and video which is quite interesting.
Visit Reference Link: https://github.com/jitsi/webrtc/tree/master/examples regularly.
As Problem states;
I want to record but smaller size. I've tried MediaRecorder and it doesn't work because WebRtc is recorded and MediaRecorder has not permission to record while calling.
First of all, to reduce or minimize the size of your recorded data (audio bytes), you should look at different types of speech codecs which basically reduce the size of recorded data by maintaining sound quality at a level. To see different voice codecs, here are well-known speech codecs as follows:
OPUS
SPEEX
G7.11 (G-Series Speech Codecs)
As far as size of the audio data is concerned, it basically depends upon the Sample Rate and Time for which you record a chunk or audio packet.
Supppose time = 40ms ---then---> Reocrded Data = 640 bytes (or 320 short)
Size of recorded data is **directly proportional** to both Time and Sample rate.
Sample Rate = 8000 or 16000 etc. (greater the sample rate, greater would be the size)
To see in more detail visit: fundamentals of audio data representation. But Webrtc mainly process 10ms audio data for pre-processing in which packet size is reduced up to 160 bytes.
Secondly, If you want to use multiple AudioRecorder instances at a time, then it is practically impossible. As WebRtc is already recording from microphone then practically MediaRecorder instance would not perform any function as this answer depicts audio-record-multiple-audio-at-a-time. Webrtc has following methods to manage audio bytes such as;
1. Push input PCM data into `ProcessCaptureStream` to process in place.
2. Get the processed PCM data from `ProcessCaptureStream` and send to far-end.
3. The far end pushed the received data into `ProcessRenderStream`.
I have maintained a complete tutorial related to audio processing using Webrtc, you can visit to see more details; Android-Audio-Processing-Using-Webrtc.
There are two parts for the solution:
Get the raw PCM audio frames from webrtc
Save them to a local file in compressed size so that it can be played out later
For the first part you have to attach the SamplesReadyCallback while creating audioDeviceManager by calling the setSamplesReadyCallback method of JavaAudioDeviceModule. This callback will give you the raw audio frames captured by webrtc's AudioRecord from the mic.
For the second part you have to encode the raw frames and write into a file. Check out this sample from google on how to do it - https://android.googlesource.com/platform/frameworks/base/+/master/packages/SystemUI/src/com/android/systemui/screenrecord/ScreenInternalAudioRecorder.java#234
I am trying to play raw sound data using AudioTrack class in Android, I am using the write method, but I noticed that there is a latency between the write method returns and the actual sound is played, to make it simple let us use AudioRecord class as the following psedu code:
//init AudioTrack
//init AudioRecord
while(true){
byte [] buffer = new byte[1000];
int read = audioRecord(buffer,0,1000);
audioTrack.write(buffer,0,read);
}
I expect to get latency that is read / sample rate seconds but the actual sound is played after and extra of about 0.5 seconds, I really need the audio to be played with minimum latency, so does anyone has an explanation of what is going on and is there any available solution or should I accept this as it is a hardware issue?
I'm assuming your goal is to come up with some interactive audio solution (that is, where sound is played in response to some user action), because in this scenario low latency really matters.
On Android, to achieve the lowest latency you need to use Open SL ES API which is available to native (C++) code via NDK. The only Java side mechanism that can achieve low latency is SoundPool class, but it has limitations in what kind of sounds you can play.
For more information, see the page on high-performance audio, and also check out this SO answer: Low-latency audio playback on Android
I am developing an Android application which is supposed to capture audio with the built-in mic of the smartphone and save it.
For the further processing purposes I would like to have some control over the quality of audio captured. For instance, to my knowledge some smartphones have high-quality audio recording mode and I would like to make use of it, if that is possible.
I am aware of mediaRecorder, but I am not sure how to use its methods or input arguments to get the best quality of sound possible. I would be very grateful if somebody could point out that for me or provide references to other libraries that allow to adjust the quality of recorded sound.
Is there any way to intercept or just-read the audio output in android device?
I need to read the whole audio output in PCM from inside myActivity, including media player application in background, voice from calls, MediaPlayer istances inside myACtivity, etc., everything that's going to played by speakers. Actually, if it was possible to read them separately, would be great as well.
I tried with AudioRecord, giving it as audioSource parameter every constant found in MediaRecorder.AudioSource with no luck, should I try different audioSources?
Is it a so low-level task that has to be handled within native layer?
The visualizer class is helpful. I use it to playback immediately played audio here.
This audio comes in very low quality, however, so it's only really good for visualization.
I am new to android and presently doing android voice recording application. I want top know which format is best for saving audio file in android. (i.e RAW-AMR or 3gp or mp4).So rhat we can hear playback sound loudly in device.
Is there any alternative way to increase audio sound through voice processing in android.
Thanks in advance.
Question: Which bear is best? Answer: Black Bear
Seriously though, you would need to state your criteria for the audio file for us to make a codec recommendation. Does it need to be portable? Best compression? Highest fidelity?
The codec that you choose has no affect on the loudness of audio that will be played over the device, so this should not factor into your criteria.
Is there an alternative way to increase audio?
Yes, if you are recording audio from the microphone then you can amplify the audio data before you save it to a file.
Let an audio sample from the microphone be represented by the function:
f(t)
Amplification is achieved by multiplying the audio sample by some factor A
A * f(t)
You can use AGC(Automatic Gain Control) module from WebRTC to increase sound level.
I didn't find any simple Java API yet. You can use C++ API via JNI.
Have a look here, WebRTC AGC (Automatic Gain Control) .
I want top know which format is best for saving audio file in android.
To save voice audio on Android (or any other platform), take a look at Opus. It's a free, state-of-the-art audio codec that also supports voice mode.