good microphone for vlogging

In the production of the publishing side, you need to collect the sound card output signal, in order to mix with the microphone input signal, for audio processing technology, mainly the following: Capture Microphone Input Collect sound card output Send audio data to the sound card for playback Mixing multi-channel audio inputs Here are two codes to refer to:A.XP DirectSound Audio a

interface is not the same as websocket, the communication between a browser and the WebSocket server, but through a series of signaling, to establish a browser and browser (Peer-to-peer) channel, the channel can send any data, Without having to go through the server. and WEBRTC through the implementation of MediaStream, through the browser to invoke the device's camera, microphone, so that the browser can transfer audio and videoWEBRTC is already in

configuration is invalid and will be ignored;4) There should be only one string node in each permission node , and the extra string nodes will be ignored;5) $description within a string node require user Configuration and can be any string that represents The Description field that is displayed to the user when the APP requests permissions. Example :To use the Uexvideo plugin to record video , microphone and camera permissions are required, then

Avaudiosessioncategoryplayback Avaudiosessioncategorymultiroute Avaudiosessioncategoryoptionduckothers Smart Low Conflict Audio volume Avaudiosessioncategoryplayandrecord Avaudiosessioncategoryplayback Avaudiosessioncategorymultiroute Avaudiosessioncategoryoptionallowbluetooth Allow Bluetooth audio input Avaudiosessioncategoryrecord Avaudiosessioncategoryplayandrecord Avaudiosessioncategoryoptiondefaulttospeaker Default outpu

are two big advantages: the semantic side, and the ease with which you can expand, not just the audio/video devices.Now breathe a deep breath. This thing is fast!3rd round: WebRTCThe elements eventually disappeared like the dodo.Relying heavily on WebRTC (network instant Messaging), the pace of finding the right capture API in recent months has accelerated a lot. The specification is supervised by the WebRTC Working Group. Google, Opera, Mozilla and other companies are currently working on impl

This morning to deal with the WIN7 system of the computer front microphone and headphone port can not be used, the main phenomenon is the headset after inserting no sound, wheat inserted into the microphone can not hear the speaker, the rear port all normal. At the beginning of the judgment is definitely set the problem, so with another computer Baidu search "Win7 computer front panel headset

such as video, its "computer screen video" Can not only be used to make tutorials video, but also record player video, game video, qq/msn chat video and so on. Many times, when we are recording tutorials or games, we need to add our own narration or commentary to make the video more contagious. Recordings include recording system sounds, microphone sounds, and recording system and microphone sounds simult

Recently updated the previously released speech recognition framework Myvoix, adding a microphone waveform analysis effect. have not seen Myvoix introduction of classmates please poke (portal) GitHub address　In the new update, waveform analysis can be tied to a microphone source, or it can be used separately with other sources. First, take a look at the code for waveform drawing of the

Internet age, the distance between time and space has long been narrowed down by the network, we can use Instant Messaging software in Win7 system and distant friends for voice chat, but sometimes they can hear each other's voice, but the other side can't hear our voice, carefully check the headset microphone also no problem, this how to do? Then we need to do some checking on the sound system configuration of the Win7 computer. First, right-click th

Now many laptop users are using the Win7 system, we know that there is a built-in microphone inside the notebook, so there is a new chat form other than typing chat-voice chat. Voice chat, the most afraid of seeing is unable to hear the voice of the problem. In fact, most of it is due to microphone settings, the following small set to teach you how to solve this type of problem. First, we use the mouse in

(! Bufferstate.buffer1full) { Break } } pbuffer = (int16_t *) bufferstate.buffer+ Audio_frame_size*bufferstate.readindex; if (waveformat.channels = = 1) { for (i=0; i I2sstate.txbuffer[i2sstate.txwriteindex][i]= ((int16_t *) pbuffer) [i]; } totalsize+= audio_frame_size * sizeof (int16_t); bufferstate.readindex++; } else { for (i=0; i I2sstate.txbuffer[i2sstate.txwriteindex][i]= ((int32_t *) pbuffer) [i]; } totalsize+= audio_frame_size * sizeof (int32_t); bufferstate.readindex+= 2; } if (Buffe

, which is widely used in Voip,phone, but is not used in ASR. It is estimated that the use of voice activity detection and comfort noise generation can reduce the bandwidth requirements of a set of audio channels by 50%. There are two commonly used DTD algorithms: energy-based, such as the Geigel algorithm, the basic principle is to detect the near-end signal strength if large enough to judge someone to speak. Based on signal correlation, some correlation algorithms, such as cosine similarity, a

Acoustic echo refers to the sound that the speaker plays out and sends back to the far end when the microphone is picked up, allowing the remote interlocutor to hear his or her voice. Acoustic echoes are also divided into direct echoes and indirect echoes. Direct echo refers to the speaker playing out the sound without any reflection directly into the microphone. The echo delay is shortest, and it is relate

, which is widely used in Voip,phone, but is not used in ASR. It is estimated that the use of voice activity detection and comfort noise generation can reduce the bandwidth requirements of a set of audio channels by 50%. There are two commonly used DTD algorithms: energy-based, such as the Geigel algorithm, the basic principle is to detect the near-end signal strength if large enough to judge someone to speak. Based on signal correlation, some correlation algorithms, such as cosine similarity, a

Transferred from: http://shanewfx.github.io/blog/2013/08/14/caprure-audio-on-windows/When a task is received in the previous period, the output signal of the sound card needs to be collected in order to mix with the microphone's input signal.before considering how to implement this requirement, let's discuss three modes of computer sound:1) Render modeThis method is actually playing (output) sound, common APIs such as PlaySound, WAVEOUTXXX, DirectSound, etc.2) Capture modeThis approach is actual

How to Set Realtek HD sound card in VISTA/WIN7, hdwin7 Now more and more people are using VISTA and win7 systems, and most of them are Realtek HD sound cards. The setup problems are quite different from those of Windows XP systems, next I will introduce some experiences in setting a sound card in the VISTA system. If you have any mistakes, please advise me. 1. Open Control Panel and enable sound option (figure 1 ). Figure 1Open the sound option in Control Panel 2. In the "sound" dia

First, open Rad Studio XE8, new project, add Components MediaPlayer, cameracomponent and some buttons, from top to next for Button1, 2, 3, 4, 5, 6, 7, 8;Second, in the header file. h add:#include , and__published:void __fastcall GetImage ();PrivateTaudiocapturedevice *microphone; Declaring a microphoneString FileName;Third, write the code in each button:Note that there is #include in the. cpp file, and the remaining code is:__fastcall Tform1::tform1 (

Under pressure, I studied red5 and as3 last night and found it quite interesting.Red5 and fms are both good things, and they find learning things very fast under pressure. The code is as follows:Copy code // Pre-defined statementImport mx. utils. Delegate;// Create a basic network connection objectVar nc: NetConnection = new NetConnection ();// Connect to the local Red5 serverNc. connect ("rtmp: // localhost/online ");// Pass the nc that has established a connection with the red5 s

Audio interfaces include analog and digital audio interfaces. Analog audio interface: (1) line interface:It is the most frequently used audio interface for connecting video and audio devices. It can be divided into professional device line interface and non-professional device line interface. Line Interface of professional equipment (balance port): used for connection in broadcast and professional audio systems. Basic features: (A) Input InterfaceRated input level: 0 dB or + 4 dBMaximum

. Play (); II. music: In silveright, The MediaElement control is also used to play mp3 music. However, it is not appropriate to use it as background music, therefore, the Song and MediaPlayer classes provided by XNA can be used to load and play background music, which is also well controlled. Song song = Song. FromUri ("Kalim", new uri(?kalimba=", UriKind. Relative )); MediaPlayer. Play (song ); 3. Microphone is used to record sound in silverlight. Us