webrtc h264

, the receiver side decoding good performance, no mosaic phenomenon.3.2, adding the QoS module will bring a certain delay and lag, because packet retransmission is time-required.3.3, the above plan is WEBRTC inside the nack concrete realization way.The above scheme is provided by Peng Zuyuan, a senior audio and video expert from the ring, with some adjustments, and Kelly for editing and finishing.Peng has many years of audio and video codec developmen

This article mainly introduces to help a programmer solve WEBRTC doubt process, the article from the blog Garden Rtc.blacker, support original, reprint please explain the source (www.rtc.help)This article mainly comes from the mail, why I will be specially organized into essays, mainly based on the following reasons:1, the author email me The purpose is to ask questions, but he asked questions in a way worthy of praise, asked very specific (if asked t

1, about WEBRTCWebRTC is a very popular project. The first problem encountered is the WEBRTC compilation problem.Fortunately, a company has helped compile and put it in Maven's repo.Address:Http://mvnrepository.com/artifact/io.pristine/libjingleThe update is very fast, and WEBRTC the official Basic sync update.2,android DemoThe project is also within the pristine project:Https://github.com/pristineio/apprtc

The combination of gstreamer and webrtc is a little breakthrough, gstreamerwebrtc Today, I found a fork killer in gstreamer, and quickly came up with a general framework and solution plan, using the gst-inspector to perform object introspection attribute detection first, then, the gst-launcher tool is used for Pipeline Test. Finally, the channel Logic Source Code is implemented using c to implement webrtc-

from a downhill racing race. Most of the video remains the same, except that the moving parts, i.e. the car and the audience, need to be encoded as P-frames without changing the video. The I frame is generated as a new reference point for P frames. Usually create an I-frame when the image changes very much, such as: panning, scene switching, a large number of actions, sudden disappearance and other scenes. error recovery mechanism:it is suitable for the error recovery mechanism of various packe

The previous article (WEBRTC Audio-related Neteq (a)) is an overview of Neteq, know that it is mainly used to solve the network delay jitter drops and other problems to improve the voice quality, but also know that it has two large units of MCU and DSP components. MCU is mainly received from the network of voice RTP packets into the packet buffer, but also based on the calculated network delay and jitter buffer delay and the feedback from the DSP unit

WebRTC (Web Real time communication) is not Google's original technology, in 2010, Google bought about $68.2 million for VoIP softwareDeveloper Global IP Solutions Company, open source WEBRTC real-time communication project.Voice engine is the gips of voice communication, it is mainly through a series of transmission control to achieve low bandwidth transmission of real-time voice, Gips speech engine hasa w

IVideo Encoding 1.1 target of video compression and encoding 1) Ensure compression ratio 2) Ensure recovery quality 3) easy to implement, low cost, and reliability 1.2 starting point of compression (feasibility) 1) Time Correlation In a video sequence, adjacent two adjacent frames have very few differences. This is the time correlation. 2) Spatial correlation In the same frame, there is a large correlation between adjacent pixels. The closer the two pixels are, the stronger the side correlation

Recently I want to upgrade my TV, so I want to buy a 46 (47) TV. I have locked several models on the Internet. Today I went to Suning and Dazhong and Gome to see the actual results, I would like to share with you that everything is my subjective feeling and the feeling of a batch of prototypes I have seen. So only one reference is provided. Source of the test: 1. 720 p MKV format x264 embedded subtitle cartoon aptx Conan. 2. h264 1080 p Deep Blue 3,

If you want to transmit video streams in real time when using Android phones for h264 hard encoding, you need to know the sequence parameter sets (SPS) and picture parameter set (PPS) of the video stream ). Today, I understand how to obtain SPS and PPS. I will record them here. I hope you can get some help here. First, let's take a look at the prerequisites. The video recording parameters I set are: Mmediarecorder. setoutputformat (mediarecorder. outp

I wrote an article earlierArticleAnalysis of the format of using RTP for h264 packets: RTP encapsulation of h264. However, it seems that the split and some situations that need attention are not clearly stated, so here we will make a supplement and also serve as our own memo (I don't seem to have a good memory ). note that the sampling rate of h264 is

To play the H264 bare stream, you can split it into the following three jobs:1. Decoding H264 bare stream to get YUV data2. Convert YUV data to RGB data fill picture3. Display the captured pictureTo complete the work 1, we can directly use the HiSilicon decoding library, because the HiSilicon decoding library is C + + dynamic library, to complete in C # call can refer to HiSilicon

slice is incremented by 2.The frame rate recorded in the SPS in H264 is usually twice times the actual frame rate time_scale/num_units_in_tick=fps*2Therefore, the actual calculation formula should be this wayPTS=1000* (I_FRAME_COUNTER*2+PIC_ORDER_CNT_LSB) * (Time_scale/num_units_in_tick)or aPTS=1000* (I_FRAME_COUNTER+PIC_ORDER_CNT_LSB/2) * (TIME_SCALE/NUM_UNITS_IN_TICK/2)So, the 11th frame of PTS should be so calculated1000* (9*2+2) * (Time_scale/num

iOS audio AAC video H264 coded push flow best practicesProjects are personal research and experimentation, there may be many bad or wrong places please forgive.1 Overview of features* Realization of audio and video data collection* Realize the encoding of audio and video data, video encoding into H264, audio encoding into AAC* To achieve the release of audio and video data, the encoded audio and video trans

, skip here.But there is a problem to note that non-IE browser session will be lost, the search for a lot of data, the final summary of the reasons are: Because Uploadify uses a flash client, it produces useragent different from the user-agent of the browser. Final Solution: Copy Code code as follows: Add the session parameter to the upmodify upload parameter as follows: Scriptdata: {"session_id": "}, Add the following code to the server-side receive page: if (@$_request['

H264 es raw data is generally in the format of the NAL (Network Abstract Layer). Can be used directly for file storage and network transport. Each nalu (Network Abstract Layer Unit) data is composed of data header +rbsp data. The first step is to split the data stream into a single, NALU data. The value of Nal_type,i_nal_type that gets Nalu equals 0x7 indicates that the NALU is an SPS packet. Find and parse this SPS packet, which contains very importa

Step by step learning, a little progress every day FFmpeg + x264 + QT Decoding code H264 Decoding: H264 encoded format of the MP4 file decoded after saving RGB to PPM format Encoding: Encode the decoded RGB format to H264 Code: Decoding section: . Pro TEMPLATE = AppCONFIG + = ConsoleCONFIG-= qt SOURCES + = Main.cppIncludepath + =-i/usr/local/include/LIBS + =-l/

I. H264 Basic Concepts 1.1, NAL, slice and frame introduction and interrelated NAL refers to the network extraction layer, which put some network-related information. Slice is the meaning of the film, 264 in the image into a frame (frame) or two fields (field), and the frame can be divided into one or several slices (slilce), the slices are composed of macro blocks (MB). A macro block is a basic unit of encoding processing. A frame can be divided into

WEBRTC Voice Overall framework Figure One voice overall frame diagram As shown above, the entire processing frame of the audio is responsible for the transmission of the peer data in addition to the Ligjingle, mainly the Voe (Voice Engine) and the channel adaptation layer Figure II Creating a data communication channel timing diagramThe image above is the local sideComplete process, Voe is created by Createmediaengine_w, the channel adaptation layer

This article original from Http://blog.csdn.net/voipmaker reprint annotated source.WEBRTC provides real-time, web-based audio and video data interoperability, but WEBRTC can also run as a native app on a mobile platform, WEBRTC is a set of media frameworks, implemented in C + +, and officially ported to mobile platforms, including Android,ios, Platform-corresponding development language can be directly deve