audio-recording

问题 I am a newbie in Android and I was trying to build an app to record audio data. I am using Eclipse Galileo IDE with ADT plugin. And my app is targetted for the Andriod 2.1 platform. Unfortunately the example provided in the Dev Guide throws many exceptions. For example: to get MIME type the code uses recorder.getMimeContentType() . But this method does not exist in my version of MediaRecorder class. I have searched online as well as this forum and came up with one or two alternatives which

问题 I have used auriotouch codes in my app and when i record the audio, it shows the audio waves . So, while recording the sound, the mic recognizes the audio input and then the waves would act accordingly to whatever the sound the mic receives. So far its fine. But now, when I click on the play button to play the sound I just recorded, the mic's input should be off, so that the waves would act only according to the audio I recorded before and the waves should not act even if I speak while it

问题 I've tried to overcome this for a while. I'm trying to record sound, but the AVAudioRecorder doesn't record while screen is locked. It does continue to record once screen is unlocked, but the audio recorded when screen was locked is lost forever. I can't find anything wrong with what I'm doing: -(void) startRecording { // Begin the recording session. _session = [AVAudioSession sharedInstance]; NSError *setCategoryError = nil; NSError *startRecordError; [_session setActive:YES error:

问题 So far, I thought it was a firmware error, but now I just found a 2nd device that has two internal microphone capsules but only yields a monophonic signal (1st one was the S2 Plus GT-I9105P with Android 4.1.2 and now the HTC One M7 ; the Nexus 10 with Android 4.4.2 has only one mic). I tested with: bufSize = AudioRecord.getMinBufferSize(44100, AudioFormat.CHANNEL_IN_STEREO, AudioFormat.ENCODING_PCM_16BIT); ar = new AudioRecord(source, 44100, AudioFormat.CHANNEL_IN_STEREO, AudioFormat.ENCODING

问题 I am making an audio recorded using NAudio in C# and i need to remove the stop button used and simply stop the recording on its own after some time delay. The code for the record event is private void cmbRecord_Click(object sender, EventArgs e) { outputFilename = "file address"; waveInStream = new WaveIn(44100,2); writer = new WaveFileWriter(outputFilename, waveInStream.WaveFormat); waveInStream.DataAvailable += new EventHandler<WaveInEventArgs>(waveInStream_DataAvailable); waveInStream

问题 I'm working on an audio recording system. My task involves extracting the SMPTE time code from audio input stream, generated by a synchronizer device. I'm using ASIO SDK to get the time code of each callback buffer but it's always zero. Perhaps somebody has experience in ASIO SDK (or any other platform/sdk that can be used to extract SMPTE timecode from audio stream) could help me? Regards, Ben 回答1: LTC is straightforward, so If nothing else, you could just scan the audio stream for LTC data,

问题 I am trying to record audio, after that should play record immediately. Where I am going wrong. Please help me out. I am not getting. showing null pointer exception... Here is the code import android.app.Activity; import android.media.AudioFormat; import android.media.AudioManager; import android.media.AudioRecord; import android.media.AudioTrack; import android.media.MediaRecorder; import android.os.Bundle; import android.util.Log; import android.view.View; import android.view.View

问题 In recorderjs rec.exportWAV([callback][, type]) is used to generate a Blob object containing the recorded audio in WAV format. But the WAV file is taking large space, for 30sec it is approximately 1mb. So I want to record the audio in format of mp3/m4a which takes less space. Is there any way to record audio in format of mp3/m4a. 回答1: i think you should check this link : HTML5 record audio to file here u can change the audio type like this : type = type || config.type || 'audio/wav'; 回答2:

问题 I am using recorder.js to record two audio files on my web page, which then creates recordings as blobs. Once I have these audio blobs I would then like to combine them both into one track to be played by another html5 audio player. I've searched the internet but I can't seem to find any documentation on combining two blob files into one. I've found how to concatenate one blob to the other but I'm wanting to merge the files at the beginning. i.e. - sound 1 is singing and sound 2 is guitar.

问题 I'm wondering if it's possible to record an audio file in mp3 format programmatically with Xcode. I've read a tons of documentations and I've googled for this, but no result. Please help, if someone knows how this is possible? 回答1: I would start here http://developer.apple.com/library/ios/#referencelibrary/GettingStarted/GS_AudioVideo_iPhone/_index.html, try to record some raw sound file and then convert it to mp3 with some 3rd party library like LAME. You can pick up LAME source here: http:/