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to ensure real-time audio transmission has become one of the top problems in VoIP technology. This article first introduces the principle and basic implementation process of VoIP, and then conducts experimental research on real-time aud
After more than two months of hard work, we have achieved initial results in real-time audio/video conversations on the iPhone 4 platform. I checked a lot of information and thanked the information provider. In the future, I will write down this article. I will only list the key points, the specific code and the platform port, and try your best. Follow these step
When using the js-sdk to collect audio, you want to display the spectrum in real time, but you do not know how to detect the recording parameters in this case. Is there a good way to achieve the desired effect like webaudiodemos. appspot. comAudioR? When using the js-sdk to collect audio, you want to display the spect
I recently learned about Streaming Media on Android. I didn't know anything at first, and I didn't know where to start. I just went online and looked at the information. It was really a matter of feeling the stone. Later, I learned a little more about the materials. I would like to thank the people who provided the materials online. I have learned a lot. Now the real-time transmission and playback of
When capturing audio with JS-SDK, you want to be able to display the spectrum in real time, but do not know how to test the recording parameters in this case. The effect you want to achieve is like Https://webaudiodemos.appspot.com/AudioR ... Is there a better way to implement it? Reply content: When capturing audio
, Buffer = The number of audio frames (48000/1024) multiplied by the time scale (300/1000) = (48000*300)/(1024*1000) = 14.0625 per second. 2 MP3 MP3 Each frame is 1152 bytes, then: frame_duration = 1152 * 1000000/sample_rate For example: Sample_rate = 44100HZ, the calculated length is 26.122ms, this is often heard mp3 each frame playback time fixed to 26ms. ===
a frame video. because this 22.32ms or 40ms is not an accurate, or the sound card broadcast time is not the same . Here you need to know how long it takes for a sound card to broadcast a frame/or to play a buf audio. 2: The sound card broadcasts a sample point instead of one frame at a time. The sound can be heard when a sample point is lost, and the video is no
updated label. Convert data into required document and index The benefit of doing so is to completely remove the index from the above and then rebuild it: Data source query scans for small amounts of data The corresponding update index has fewer bars, reducing the number of IndexWriter commit and close these time consuming operations The above solves the problem of increment, but the re
file or the data in the data source with the updated label. Convert data into required document and index The advantage of doing so is to completely remove the index from the above and then rebuild it: Data source query scans for small amounts of data The corresponding update index has fewer number of bars. Reduced a lot of indexwriter commit and close these time-consuming operations The above overcomes the problem of incre
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.
Effect Display: http://k.thec.cn/liuping2006/music_jump_line/music_jump_line.swf If you are interested, I will introduce the method of making. The production of real-time audio beat barFlash's processing of sound mainly includes:In the current version of Flash (Flash5, FLASHMX, Flash2004, Flash8), flash on the sound file processing capacity is not strong enough.
thatMaintains and updates the flight plans for each cial airliners. These can operate to a latency of seconds. Live audio-video systems are alsoUsually soft real-time; violation of constraints results in degraded quality, but the system can continue to operate. It is important to note that hard versus soft real-
ffmpeg service implementations, please refer to: http://blog.csdn.net/eguid_1/article/details/51777716 http://blog.csdn.net/eguid_1/article/details/51787646 can also refer to the JAVACV Converter implementation: JAVACV Development of the 4: the converter implementation, implementation rtsp/rtmp/local file forwarding to the RTMP server Java Package ffmpeg command, support native ffmpeg all commands, Realize ffmpeg multi-process processing and multi-threaded output control (open, close, query),
Reprint Please specify source: http://blog.csdn.net/xiaojimanman/article/details/43982653Http://www.llwjy.com/blogd.php?id=63d4c488a2cccb5851c0498d374951c9Personal blog Station also build success, website: www.llwjy.com/blog.php, Welcome to spit Groove ~Basic principleIn the previous blog also said that the program initial session index file is very consumption of system resources, so in order to achieve real-time
In fact, as early as June 2 ago, a friend who worked at Google told me this information. I also got all the source code from WebRTC for the first time, but since my recent work was really busy, this information was not immediately reproduced here. Now, I want to keep an eye on the multimedia applications, learn the technologies in the WebRTC Framework earlier, and use them in actual projects. Google today announced the opening of the WebRTC architec
Streaming media refers to the continuous time-based media transmitted using stream technology in the network. It features that the entire file does not need to be downloaded before playback, but is played by downloading and playing, it is the technical basis for video conferences, IP phones, and other applications. RTP is a standard protocol and Key Technology for Real-
Transferred from: http://www.aiweibang.com/m/detail/104476372.html?from=pWith the gradual popularization of WEBRTC standard, real-time audio and video communication technology has been paid more and more attention by companies and technicians. For interactive audio and video applications, stability, low latency, and cl
: the status of the current index. There are two statuses: sleeping and working. The so-called sleeping means that newly added documents only enter the memory index, the so-called working means that one of the memory indexes is being merged with the index on the hard disk. We will discuss it in detail in the next section of the real-time mechanism. The mem _ mem structure is the key to using two memory ind
means that newly added documents only enter the memory index, the so-called Working means that one of the memory indexes is being merged with the index on the hard disk. We will discuss it in detail in the next section of the real-time mechanism. The Mem _ mem structure is the key to using two memory indexes and one hard disk index to implement real-
live stream and push it (you can save it on the live side) rtmp and HLS are the mainstream streaming media protocols in the Web world. The use of RTP or RTSP protocols is generally monitored. Streaming Media protocol selection: RTMP based on TCP protocol, RTMP can maintain a delay of about 3 seconds. HLS is based on the HTTP protocol, so the real time is very poor, want to use HLS to keep
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