audio

问题 I'm new to soundio. I'm wondering how to play more than one audio source at once. I'm looking to understand if I'm supposed to create several streams (seems wrong) and let the operating system do the mixing, or am I supposed to implement software mixing? Software mixing seems tough too if my input sources are operating at different frequencies. Am I basically asking "how to mix audio"? I need a little direction. Here's my use-case: I have 5 different MP3 files. One is background music and the

问题 I need to play a sound (.WAV file) from a Visual Basic 6 application. However, the PC is equipped with multiple sound cards and the user should be able to use it. Up till now a default sound card was supported and I would play a sound like this: CommandString = "Open " & Chr$(34) & mstrFilename & Chr$(34) & " type waveaudio alias MediaFile" mciSendString CommandString, vbNullString, 0, 0& If PlayFromPercent <= 0 Then CommandString = "play MediaFile" lngRetcode = mciSendString(CommandString,

问题 I am doing some preparation research for an Android phone app that would involve voice search or speech recognition while the user is simultaneously engaged in a phone call. I'm finding that when I am in the middle of a call and attempt to invoke voice search or a similar action that would take input from the microphone I get an "audio error" message. Does the phone app take exclusive control over the microphone input of the phone such that other apps cannot get access to audio input? 回答1: In

问题 I am doing some preparation research for an Android phone app that would involve voice search or speech recognition while the user is simultaneously engaged in a phone call. I'm finding that when I am in the middle of a call and attempt to invoke voice search or a similar action that would take input from the microphone I get an "audio error" message. Does the phone app take exclusive control over the microphone input of the phone such that other apps cannot get access to audio input? 回答1: In

问题 Some .wav file I can't play with soundpool. I can't hear anything. Some files play just fine. Why? code AudioManager mgr = (AudioManager) context.getSystemService(Context.AUDIO_SERVICE); int streamVolume = mgr.getStreamVolume(AudioManager.STREAM_MUSIC); soundPool.play(soundPoolMap.get(sound), streamVolume, streamVolume, 1, 0, 1f); 回答1: I had lots of problems with soundpool and file formats. Read my original question, it might be helpful to you. Basically, I changed to use MediaPlayer and had

问题 Some .wav file I can't play with soundpool. I can't hear anything. Some files play just fine. Why? code AudioManager mgr = (AudioManager) context.getSystemService(Context.AUDIO_SERVICE); int streamVolume = mgr.getStreamVolume(AudioManager.STREAM_MUSIC); soundPool.play(soundPoolMap.get(sound), streamVolume, streamVolume, 1, 0, 1f); 回答1: I had lots of problems with soundpool and file formats. Read my original question, it might be helpful to you. Basically, I changed to use MediaPlayer and had

问题 I have a VXML (say vxml-1) which contains simple audio file for announcement followed by a goto element pointing to next vxml doc(say vxml-2) containing another audio file for announcement. Once the execution starts, vxml-1 loads with corresponding audio followed by goto execution. Now when the user hangs up, in the background it shows that it ended at vxml-2 because of very fast goto transition, but actually the user hung up at the audio of vxml-1 because of its long length. How can I

问题 I have a VXML (say vxml-1) which contains simple audio file for announcement followed by a goto element pointing to next vxml doc(say vxml-2) containing another audio file for announcement. Once the execution starts, vxml-1 loads with corresponding audio followed by goto execution. Now when the user hangs up, in the background it shows that it ended at vxml-2 because of very fast goto transition, but actually the user hung up at the audio of vxml-1 because of its long length. How can I

问题 I am trying to send an OGG_OPUS encoded stream to google's speech to text streaming service. Since there is a time limit imposed by Google for their stream requests, I have to route the audio stream to another Google Speech To Text streaming session on a fixed interval. From what I've read, the pages in the OGG stream cannot be read independently since the data in the pages are calculated by considering the data of the previous and next pages. If that is the case, can we cut off the stream at

问题 Closed . This question needs to be more focused. It is not currently accepting answers. Want to improve this question? Update the question so it focuses on one problem only by editing this post. Closed 2 years ago . Improve this question First time working with audio in python, found many examples where you can listen/stream microphone, but with a lot of googling, can't find how can I listen just my computers output audio. The idea is to listen audio stream for comparing the sounds to wav.