bogen pcm

This article is free part of http://blog.csdn.net/droidphone1. What is PCM? PCM is the abbreviation of Pulse-code modulation. We know that in real life, the voice heard by human ears is a analog signal. PCM is a technology that converts the voice from analog to digital signal, his principle is simply to use a fixed frequency to sample the analog signal. The sam

Scenario One: Use Avaudiorecorder to record the PCM data format file, and then start the sub-thread loop through the file, read the PCM data transcoding MP3 and write to the mp3 file (most of the current online information is like this)1.lame Initialize Open PCM, MP3 file get file pointerSingle channel does not need to set Lame_set_model, because API has been de

Like videos, audio transmission often requires compression. The following describes the use of the PCM audio bare stream File compression and decoding library provided by hith, comparing the compressed and decoded data of a file, we can clearly find that the audio compression process of PCM> ADPCM is lossy. For file comparison, I am using the BCompare software, which is definitely a must-have for programmer

Multi-channel lpcm: the original format of lossless audio tracks. It is equivalent to a wave file and does not require decoding. You can directly input a Power Amplifier for DA conversion. The fiber optic and coaxial interfaces can only transmit two-channel lpcm, multi-channel lpcm requires HDMI interface transmission. PCM: nonlinear pulse Coding Modulation Lpcm: linear pulse Coding Modulation They are an encoding method for converting analog speech s

Example 8: DirectSound PCM and directsoundpcmThis document records the techniques used by DirectSound to play audio. DirectSound is the most common Audio Playback Technology in Windows. Currently, most audio playback applications use DirectSound. This article records an example of using DirectSound to play PCM.Note: A dear friend has reminded me that DirectSound is planned to be replaced by XAudio2. Later, I found that this was the case. Therefore, in

PCM file: Analog audio signal via analog-to-digital conversion (A/D transform) directly formed binary sequence, the file has no additional file headers and file end flags. The Windows Convert tool converts files in PCM audio format to files in Microsoft WAV format. Brief introduction of pulse coded modulation PCM file format Digital audio, in fact, is to digit

Directsound capturing microphone PCM Data encapsulation class There are many ways to read PCM data from a microphone on the Internet. I just released a directsound microphone PCM Data Collection class I wrote here. Through this class, we can easily use directsound technology to collect microphone data, developers do no

How to encapsulate pcm data into wav Files in C Language Pcm is the raw audio data. wav is a common audio format in windows. It only adds a file header to the pcm data. // WAVWriter. cpp: defines the entry point of the console application. // # Include "stdafx. h" # include Using namespace std; typedef struct WAVE_HEADER {// RIFF char chunkID [4]; // long i

Tool encapsulation for PCM conversion MP3 and pcmmp3 Tool encapsulation for PCM conversion MP3 Description 1. Simple encapsulation of PCM to MP3. 2. Use the https://github.com/wuqiong/mp3lame-for-iOS to generate a 64-bit lame library. Source code Https://github.com/YouXianMing/iOS-General-Tools in PCM-to-MP3 //// Pc

PCM encoding (original Digital Audio Signal Stream) Type: Audio Maker: ITU-T Required bandwidth: 1411.2 kbps Feature: sound source information is complete, but redundant Advantages: sound source information is fully saved and sound quality is good Disadvantages: Large Information volume, large Redundancy Application: VoIP Royalty: free Note: in computer applications, PCM encoding can reach the high

Overview of PCM File Format PCM file: the analog audio signal is directly formed by Analog-to-analog conversion (A/D conversion ).Binary SequenceThe file does not have an additional file header and end mark. Windows Convert can Convert PCM audio files to Microsoft WAV files.Number of channels,Number of sampling digitsAndSampling frequency. Sampling frequency:Tha

This article mainly introduces the use of WINDOWSAPI implementation of the PCM audio method, very practical a function, the need for friends can refer to the nextThis paper introduces the method of using WINDOWSAPI to realize the playback of PCM audio, which is similar to the principle of using WINDOWSAPI to get audio recording, so it is no longer detailed to describe the parameters of the specific function

24-bit PCM 24-bit linear PCM is stored in blocks. Each block is divided into two parts. The first part contains the most significant two bytes of each channel for two samples in big endian order: The second part contains all least significant bytes of each channel for the two samples in the same order: The complete block looks like this: T = Top byte = bits 23 .. 16 M = middle byte

PCM data recorded with a microphone, 16bit, 48KHz, mono, and I would like to get the 16KHz sampling rate of the PCM data, then by reducing the sampling rate of the method to achieve 48000HZ to 16000HZ sample rate conversion. The conversion principle is relatively simple, 48000HZ to 16000HZ, actually dropped 3 times times, in the same time unit interval, 48000HZ sampled 3 points, 16000HZ sampled a point, tha

This article describes one of the simplest audio encoders based on FFmpeg. The encoder realizes the PCM audio sample data encoded as AAC compressed encoded data. The encoder code is very simple, but each line of code is very important. By looking at the source code of this encoder. The ability to understand the FFmpeg audio encoding process.This program uses the latest version of the class library (compile time is 2014.5.6). The development platform i

PCM The abbreviation of Pulse-code modulation. (I2S is only a branch of PCM, and the interface definition is the same. The sampling frequency of I2S is generally 44.1khz and 48 khz, And the PCM sampling frequency is generally 8 kHz and 16 kHz. There are four groups of signals: Bit Clock signals, synchronous signals, data input, and data output .) Two important in

;}waveformatx;typedef struct{Char chriffid[4];int nriffsize;Char chriffformat[4];}riffheader;typedef struct{Char chfmtid[4];int nfmtsize;Waveformat WF;}fmtblock;Wave Audio sampling frequency is 8khzNumber of audio sample units = 8000*0.02 = 160 (determined by sampling frequency)Number of channels 1:1602:160*2 = 320BPS decision Sample sizebps = 8--and 8-bit unsigned char--16-bit unsigned shortint Encodewavefiletoamrfile (const char* pchwavefilename, const char* pchamrfilename, int nchannels, int

My program was changed based on the faac example in the frontend directory in the faac 1.28 library. The following is the procedure of running the program: First, call faacenchandle hencoder = faacencopen (samplerate, channels, samplesinput, Maxbytesoutput ); 1.Open the AAC encoding engine and create an AAC encoding handle. The samplerate parameter is the sampling rate of the audio PCM stream to be encoded, and channels is the number of channels of

This article describes one of the simplest audio encoders based on FFmpeg. The encoder realizes the PCM audio sampling data encoded as AAC compressed encoded data. The encoder code is very simple, but each line of code is important. By looking at the source code of this encoder, you can understand the FFmpeg audio coding process. This program uses the latest version of the class library (compile time is 2014.5.6), the development platform for VC2010.

# Include "ALSA/asoundlib. H" // 11 kHz support, longer audible time, slower sonic speed# Define sample_rate 48000// # Define sample_rate 11000 // It is indeed a two-channel Interaction# Define Channels 1// # Define channels 2 // If latency is too small, it will cause snd_pcm_writei () to lose voice data, resulting in the short write Phenomenon# Define latency (1000000) // 1sec # Define nblocks 16# Define block_size 1024Unsigned char buffer [nblocks * block_size];/* here: some random daTa: Futur