* inputbuffer, unsigned int samplesinput,unsigned char *outputB uffer,unsigned int buffersize);//Description: Encode a frame information//hencoder:faacencopen The encoder handle returned//inputbuffer: input information buffer//samplesinput: Faacencopen encoded data length, that is, buffer length//outputbuffer: Encoded output information buffer//buffersize: Output information lengthint Faacapi faacencgetversion (char **faac_id_string, char **faac_copyright_string);//Description: Get FAAC version
(1) Application background(2) Principle of main noise reduction algorithm(3) algorithm flow(4) Algorithm implementation(5)------------AUTHOR:PKF-------------------time:2-6---------------------------qq:1327706646(1) Application backgroundBased on 8148 of audio ALSA acquisition, A8 encoded into g711 or DSP encoded into AAC, then RTP broadcast VLC play, sampling rate is 16000hz, 8bit, Europe and the United States Ulaw, Mono. Due to voltage or other causes of the acquisition of
[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
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Audio-visual data Processing Primer series articles:
Getting started with visual audio data processing: RGB, YUV pixel data processing
Getting Started with AV data processing: PCM Audio sampling data processing
Getting Started with AV data processing: Analysis of video stream in H.
Getting Started with AV data processing: AAC audio bitstream parsing
Getting Started with AV data processing: FLV Enca
MPlayer troubleshooting (AO_ALSA] UnabletofindsimplecontrolPCM, 0) -- Linux general technology-Linux technology and application information. The following is a detailed description. To enable Fedora 8 to play files in rmvb, mp3, and avi formats, download the corresponding decoder from the mplayer website and decompress it to the/usr/lib/codecs/folder, for 32-bit hosts, but when you play a file, find the file and open it. A window like the following may appear, and a red warning sign will pop up
This article corresponds to the opening link of the previous "C ++ playing audio stream (PCM bare stream)". It describes how to use C ++ to implement the recording function. Similarly, you can directly create a win32 console program and copy the code to a file name to use it. You can also download the project you uploaded. This project was written in VS2013, if it is a lower version, it will be okay. Click Open Link
The Code is as follows:
# Include
Record PCM lock information in GRDIt is located in the SGA of each instanceHowever, each instance only contains part of GRD, And the GRD of all instances is a complete GRD.PCM lock has three main attributes: mode, role, and PI.① Mode-- S-- X-- NullFor S and X lock modes, we will not repeat them hereNull indicates that the corresponding memory space can be reused. Before being reused, the instance cannot access the data here.② RoleRole is used to descr
;waveoutprepareheader (hwo, WH, sizeof (WAVEHDR)); //prepare a waveform data block for playback waveoutwrite (hwo, WH, sizeof (WAVEHDR)); //plays the second function in audio media WH specifies the data WaitForSingleObject (wait, INFINITE); //is used to detect the signal state of the Hhandle event, the thread is temporarily suspended when the function is called in a thread, and if, within the suspended infinite milliseconds, the object waiting by the threads becomes signaled. The function immedi
This paper introduces the method of using WINDOWSAPI to play PCM audio, which is similar to the principle of using WINDOWSAPI to obtain recording audio in the previous article, where the specific function and structure parameters are not detailed, the same parts are omitted, and only the main functional part code is introduced. As shown below:
1. Waveoutgetnumdevs
2. Waveoutgetdevcaps
3. Waveoutopen
callback function:
void CALLBACK Playcallba
Project and Source Address: Https://github.com/EasyDarwin/EasyAACEncoder
Easyaacencoder is easydarwin Open Source Media Services team to organize, develop an audio transcoding to AAC tool Library, currently supports G711A/G711U/G726/PCM and other audio format transcoding, cross-platform, support windows /linux/arm. Invoke Example
testeasyaacencoder: AAC transcoding via EASYAACENCODERAPI to g711a/g711u/g726;
Windows compilation method,
Visual Studio 20
The wave file format is part of Microsoft's riff specification for storing multimedia. A riff file starts with a file header and then a series of data blocks. A wave file is often made up of only one wave block, which contains a format-formatted FMT block and a data block that stores the sampled information. Standard wave file format
Offset
Length (bytes)
Variable name
Describe
Note
0
4
Chunkid
ASCII character "RIFF", big-endian form
In a previous article, I introduced how to use libmad in Linux to decode and play MP3 files, compile the source code, and play MP3 files normally, however, in the project, MP3 files need to be decoded into PCM files. Therefore, you need to modify the previous code based on the original code, so that you can test it.
/** This program is improved from minimad. For more details, see minimad. c ** myminimad. c, 2009/12/25, Sichuan University, China ** Com
Audio codec and real-combat article (1) PCM go to AAC (AAC encoding)Author: Liu · Poechant Blog: blog.csdn.net/poechant e-mail: zhongchao.ustc@gmail.com Date: April 7th, 2012
This uses FAAC to implement AAC encoding. 1 Download and install FAAC
The installation process here is implemented on MAC and Linux, and windows can be similar to the reference.
wget http://downloads.sourceforge.net/faac/faac-1.28.tar.gz
tar zxvf faac-1.28.tar.gz
cd faac-1.28
. C
Nickname: Marine fish 1001 Date: October 25, 2014-October 31, 2014
1. Task completed this week: noise suppression for audio recording. pcm+h264 package MP4, last week for audio g711a, and the parameter is not ideal for synthesis.
2. Task not completed this week: Timestamp issue
3. Next week plan: Find timestamps related literature
4. Key technical Point Description:
Platform: Linux HI3515
Cross compiler: Arm-hisiv100nptl-linux
The noise suppression o
Fatal error: malformed or writable upted AST file: 'unable to load module Darwin. pcm problem solving, too upted
The xcode5 compilation project encountered the following problem by accident:
Fatal error: malformed or your upted AST file: 'unable to load module "/Users/gapple/Library/Developer/Xcode/DerivedData/ModuleCache/PNBJ8YUNV3UC/Darwin. pcm ": module file not found 'note: after modifying syste
Recently in making a voice changer app, which used the PCM format to wave format, the following posted source code, hope that with the needs of children's shoes help!!! This is written in the C + + language, or it can be implemented in Java. Of course, Java calls the native function to use the JNI technique. Specific JNI technology to find information on the Internet itself.CPP file Pcm2Wave.cpp#include #includestring.h>#include#include"wave.h"#includ
The multimedia is suspended. Let's talk about a few irrelevant theme words. Although it can behave like nothing, it is quite lost. I have never believed that scores can be used to measure a person's level or something, but I still have a bottom line on my requirements. The bottom line is that I stick to my own style of capital. Over the bottom line, all justified reasons will turn into the arrogant requirements of the 80 s. Why have I fallen to the point where I am not involved? How can I spend
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