g711

Summary of the online G711 codec, it has been to the Andorid system, through the JNI to G711 and PCM, because of the two formats do not have in-depth understanding, so if you want to understand them can Baidu, here is just to achieve such a function. G711.C contains encoding and decoding methods http://www.easydarwin.org/article/Streaming/38.html Reference Artic

For example, to implement this function, the latest test tool should summarize the following: First, let's talk about the differences between E1 and T1: E1 is widely used in China and Europe. T1 is widely used in the United States and Japan. E1 = 2.048 Mbit/s = 32 * 8bit * 8000 T1 = 1.544 Mbit/s = (24*(7 + 1) + 1) * 8000 T1 uses one of the eight bits as the control bit, and after 24-channel transmission, A 1-bit control bit is also attached, so (24 path * (7 + 1) + 1) = 193 bits Becau

(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 cause

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

Bandwidth = package length × packets per second= Package length × (1/package cycle)= (Ethernet header + IP header + UDP header + RTP Header + payload) × (1/packaging cycle)= (208bit + 160bit + 64bit + 96bit + payload) × (1/package cycle)= (528bit + (

Understanding of bandwidth used by voice calls, voice call bandwidth Sometimes the customer is concerned about how much M of bandwidth to apply for to ensure the number of voice calls that the server can achieve. Here, I will only describe my personal understanding. The following uses G711 encoding as an example to describe it. At the same time, we should know that the standard encoding rate of G711 encodi

Sometimes the customer is concerned about how much m of bandwidth to apply for to ensure the number of voice calls that the server can achieve. Here, I will only describe my personal understanding. The following uses g711 encoding as an example to describe it. At the same time, we should know that the standard encoding rate of g711 encoding is 64 Kbps. In addition, it is assumed that the LAN uplink devic

three examples, of which Edge_ Detection (edge detection) used to c6accel Lib, focusing on DSP algorithm porting, not very classic, we do not introduce, decode is also relatively simple, is how to call H264 decoding, MPEG4 decoding, JPEG decoding, audio g711 decoding. We still choose Encode introduction, see encode how to record h264 video files and g711 audio files (while modifying can support MPEG4 or JP

The audio module consists of four sub-modules, which include input, audio, audio, and audio decoding. Audio input and The output module realizes the audio input and output function through the control of the HI35XX chip SIO interface. The audio encoding and decoding module provides audio codec functions in the G711, G726, and ADPCM formats, and supports recording and playback of original audio files in LPCM format. The audio input and output interface

1. snd_pcm_open: Open the handle. 2. configuration parameters: snd_pcm_hw_params_alloca, snd_pcm_hw_params_any, parameters, parameters, snd_pcm_hw_params_set_access, parameters, parameters, and snd_pcm_hw_params. 3. read/write: snd_pcm_writei and snd_pcm_readi. Note: 1. Create a handle Based on the functions to be implemented. The snd_pcm_open parameter snd_pcm_stream_capture corresponds to snd_pcm_readi, And the snd_pcm_stream_playback corresponds to snd_pcm_writei. 2. Configure parameters

source of the net charge, by the CC field specified number, up to 15; B. Message is Net charge Test point The main fields of interest are serial numbers sequence number and time stamp timestamp, A. Sequence number is illegal, that is, a series of RTP packets with an individual packet sequence is illegal, the product is discarded and normal recovery signal; B. RTP packet disorderly sequence namely sequence number disorderly order, NP is correct sort, the signal recovers correctly; C. When the

America and Japan, and the other isA-law Algorithm, mainly used in Europe and other parts of the world. Among them, the latter is specially designed to facilitate computer processing.2 Other instructions for editing G711 G723 G729 line account for how much bandwidth problemBandwidth = packet Length x packets per second = packet length x (1/pack cycle) = (Ethernet head +ip Head +udp Head +RTP Head + payload) x (1/pack cycle) = (208bit +160bit+64bit+9

Recently, I have never been in touch with anything related to voice communication, So I went around a lot of lines, especially voice compression and playing. I found the content related to direct sound playback and compression on the Internet, but the playback time is always delayed, and the compression effect is not very good. Later I found an article by Zhu 'er.Article(Http://zhuer.info /? P = 24). The timer mechanism is used to write data to secondarybuffer. In this way, the timer method is

(1) SDP description format(2) SDP example(3) SDP(1) SDP description formatM=video 1234 RTP/AVP 96a=rtpmap:96 H264A=framerate:15C=in IP4 192.168.0.104Above is a self-written RTPM=audio 1234 RTP/AVP 0a=rtpmap:0 PCMA/8000/1A=framerate:25C=in IP4 172.18.168.451.m= is the beginning of a media-level session, Audio: media type; 1234: port number; RTP/AVP: transport protocol; Payload format in 0:RTP header2.a=rtpmap: proof is a further description of dynamic binding; payload format in 0:RTP header; PCMA

Why should I parse the FLV format? In the live project encountered need to statistics Flash video frame interval length, first frame, GOP, and other key data, the inevitable need to parse the FLV file. noun definition First frame: Refers to the user to see the first video frame.First frame length: Refers to the user to open the Web page to see the first video frame.I frame: Video keyframe, contains all the graphic information.P Frame: Video interval frame, is based on the P frame before the P

we can only use SIP terminal as an extension to do the test, so that each machine needs to be equipped with a SIP account. We have regulated the following:Machine SIP account E1 card numberA (te210p) 00000000 02100000000B (te110p) 1111111111111112 051011111111051011111112The SIP and dial plans are configured separately below.Configuring SIP modules and adding SIP accountsWithin the/etc/asterisk/sip.conf file on Server B, the bindport=5058 in [general] can change the listening port. At the end o

1. sipdroid \ SRC \ org \ zoolu is the implementation of the SIP protocol stack. 2. sipdroid \ SRC \ org \ sipdroid is the implementation of softphone 3. Implementation related to stun in sipdroid \ SRC \ com 4. sipdroid the default encoding format is G711-A rate. 5. The program compiled directly using ant debug supports only the-rate and U-Rate Audio Encoding formats. Other programs must be imported using the ndk method before they can be used. 6. If

Media Proxy, disallow = All indicates that all media types of the caller are prohibited. Allow = ulaw indicates that u law encoding of g711 is allowed, and disallow and allow are or.　　　Create an account of 00000000 on server A. All other configurations are the same.Configure a dial-up planEdit the/etc/asterisk/extensions. conf file on server B and add:Exten => _ 021., 1, setcallerid (051011111111)Exten => _ 021., N, dial (Zap/G1/$ {exten: 0 })Exten =

Whiteboard Siplite sip softphone with g729 codec. Sipps free: Free Lite version of sipps Sjphone: SIP, H.323. windows, Linux, Mac OS X, and Pocket PC. Tekphone: VOIP (SIP) softphone for Windows. Telesis xphone: VoIP softphone for Telesis ip pbx systems. The applied VoIP protocol is Telesis xsip (Extended SIP) Protocol Tpad-the global VoIP network free PC VoIP sip softphone/dialler for Windows 98/ME/2000/Vista, SIP softphone supports STUN (stun.tpad.com) with choice of