[Concept]
The PCM (Pulse Code Modulation) Audio Encoding format is also calledPulse Coding Modulation. It is to convert analog signals such as sound into a symbolic pulse column, and then store records.
Principle]
A continuous analog signal with continuous values is transformed into a discrete time. The Discrete digital signal is then transmitted in the channel. pulse coding modulation is the process of first sampling the analog signal, then quantify
The first three articles have explained in detail how to convert the WAV audio format to the FLV format. This article describes how to convert the last part, first, the AMR is converted into WAV format, and then the WAV format is encapsulated
FLV format. The key part of this article is the part in the while loop. The other part can be skipped. The function d_if_decode (St, serial, synth16, _ good_frame ); the function is to encode a frame of AMR data
Recording audio in windows and recording audio in windows
Reference:
1: Open Source Code on github: Click to open the link. Because you need to write your own file header to store audio, the file storage function is used for reference here.
2: A very useful post code: Click to open the link, although a little flaw, but there are answers below.
3: A descriptive bl
I learned how to process images from the Internet and detected images with fake extensions: {code ...} is there any method similar to getimagesize for detecting audio and video, MP3 and MP4. I learned how to process images from the Internet to detect images with fake extensions:
// Detect forgery public function checkFlag () {$ this-> flag = @ getimagesize ($ this-> fileInfo ['tmp _ name']); if ($ this-> flag) {return true;} else {$ this-> error = "p
Example
The code is as follows
Copy Code
Audio.play (); playAudio.pause (); pauseAudio.volume Volume (sound value can be taken between 0-1)Audio.currenttime > 10; current time to 10 seconds (available to control forward rewind) function Go () {//forwardAudio.currenttime + 10;Audio.play ();}function back () {//RewindAudio.currenttime-= 10;Audio.play ();;}CONTROLS://Show or hide user control interfaceAutoPlay://Media is automatically playedLoop://Media loop playbackCur
1, open audacity, if pop-up "Start no longer show" before the check, "OK" can be. As shown below
2, OK now in the Audacity interface we click "File"-"open" to the computer you want to process the file import.
3, click "Open" as shown below.
4, in the lower left corner of the window "Engineering sample Rate" Right click on the black inverted triangle and pull down, select the appropriate value can be.
5. Click "File"-"Export" and click Save Address.
6, in
Mobile phone audio and video 12-display lyrics, audio and video 12-display lyrics1 _ create LyricShow
1 _ create LyricShow: text size, anti-aliasing, center alignment
Public class LyricShow extends TextView {private ArrayList
2 _ draw the lyrics-current sentence-Prefix-last sentence
$(window).bind('beforeunload', function(){ if(localId) { wx.stopVoice({ localId:localId }); }});
Leaving the current page cannot call the Wx.stopvoice interface? How to do to leave the page when audio also stops playing
Reply content:
$(window).bind('beforeunload', function(){ if(localId) { wx.stopVoice({ localId:localId }); }});
Leaving the current page cannot call the Wx.stopvoice int
Mobile phone audio and video 10-music list, audio and video 10-list
/*** Get data from local sdcard * // 1. traverse sdcard with the suffix * // 2. obtain the video from the content provider * // 3. if it is a 6.0 system, dynamically obtain the permission to read sdcard */private void getDataFromLocal () {new Thread () {@ Overridepublic void run () {super. run (); // isGrantExternalRW (Activity) context);
In the audio system, the call flowchart of the getoutput function for setting the sound output device is as follows:
There is only one derived class for audiopolicyclientinterface, that is, the audiopolicyservice class. The base classes of audiopolicyservice are defined as follows:
Class audiopolicyservice: Public bnaudiopolicyservice, public audiopolicyclientinterface, public ibinder: deathrecipient {
Therefore, the audiopolicymanagerbase
1, we have to install watermelon audio-visual software, Android system can be downloaded online and iOS devices can be downloaded to the App Store, and then open the watermelon audio and video to find the film you want to see, such as we find "I go to school" as an example
2, click on the film to enter after we click on the upper right corner of the download, and then see the download video can
This is a streamlined code that can support a very limited format.
Copy Code code as follows:
Package com.hongyuan.test;
Import Java.io.File;Import java.io.IOException;
Import Javax.sound.sampled.AudioFormat;Import Javax.sound.sampled.AudioInputStream;Import Javax.sound.sampled.AudioSystem;Import Javax.sound.sampled.DataLine;Import javax.sound.sampled.LineUnavailableException;Import Javax.sound.sampled.SourceDataLine;Import javax.sound.sampled.UnsupportedAudioFileException;
Based on the analysis in the first three articles, you only need to write a MAKEFILE file to compile main. cpp and aaccodechelper. cpp, as shown below:
Makefile:
Cxx = g ++
Objects = Main. O aaccodechelper. oLibs =-L/usr/local/lib-lfaac-wl,-rpath,/usr/local/lib-I/usrWavtoaac: $ (objects)$ (Cxx) $ (objects)-O darling $ (libs)Aaccodechelper. O: aaccodechelper. cpp aaccodechelper. h typedef. h$ (Cxx)-C aaccodechelper. cpp $ (libs)Main. O: Main. cpp aaccodechelper. h$ (Cxx)-C main. cpp $ (libs)Cl
1, as shown below if this resource is not right, you can search for other resources Baidu to try, because it is possible that the problem of the chip source.
2, audio and video pioneer software we better be the latest version, the old version can be technically less than the new version.
3, the speed problem, with 360 security guards or other speed measurement software, testing
4, to see if the problem is set, we open the software
methods in the Videoengine class. It also lists the APIs which is required to enable file tracing and/or traces as callback messages
(3), Audio Engine (voiceengine) module APIsThe following table lists the sub APIs that are currently available in Voiceengine
Sub-api
Header file
Description
Voeaudioprocessing
Voe_audio_processing.h
Adds Support for Noise Suppression
G711 Encoding
G711It is an international Telecommunication Union ITU-T Custom-made set of speech compression standards, which represent the logarithmic PCM (logarithmic pulse-code modulation) sampling standard, mainly used for telephones. It mainly uses the Pulse code modulation to the audio sampling, the sampling rate is 8k per second. It uses a 64Kbps uncompressed channel to transmit voice signals. The compression rate is 1:2, which is to compress 1
= audiorecord.getminbuffersize (8000,
Audioformat.channel_in_mono,
Audioformat.encoding_pcm_16bit);
2.Android echo Cancellation There are probably two types of echo cancellation on the Internet: 1. The audio is automatically removed from the voice_communication mode of Android. 2. Use third-party libraries for elimination (WEBRTC, Speex. ) to eliminate the echo. With third-party words, more trouble, and not good to achieve, with the ef
First 3ArticleI have implemented some functions of the audio chat room, including text chats, shared whiteboards, and voice chats. This article describes the technical points of video session implementation.
In silerlight4, the camera and sound collection functions have been integrated, but there is no directly usable control in wpf4. It can also be seen that the DesktopProgramThe general trend of web programs. If you want to use Silverlight to impl
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