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 encode
This article is mainly about the Raytheon analysis of AAC, the article links are as follows:Getting Started with AV data processing: AAC audio bitstream parsinghttp://blog.csdn.net/leixiaohua1020/article/details/50535042 Introduction:
The decoding of AAC files is that the AAC stream is decomposed into
Audio output
1-Way
Audio Compression Standard
Aac
Audio sample Rate
8k/16k/32k/44.1khz
Audio bitrate
16k/32k/96k/128kpbs
Video parameters
Video input
All the way SDI
Local video ring out
All the way SDI
Network Video output
Up to 1080P
Video Capture Signal
1. FDK-AAC
Http://sourceforge.net/projects/opencore-amr/files/fdk-aac/
The corresponding msvs version is available on GitHub:
Https://github.com/nu774/fdkaac
"Small record: The FFMPEG compiled by Intel compiler is added with support for FDK-AAc. The link is the fdkaac static library generated by vs2010, which is generated for execution.ProgramThe validity r
AAC V.s. MP3Http://en.wikipedia.org/wiki/Advanced_Audio_Coding#AAC.27s_improvements_over_MP3AAC Patent Lisense FAQ:Http://www.vialicensing.com/licensing/aac-faq.aspxNeed a license but usually no fees, exept that's when end user products is (/contain?) encoders/decoders.Overview:$ (ndk_root)/docs/opensles/index.html
Decode Audio to PCM
Note:this feature is a
It took me nearly three weeks to implement HTTP live streaming on the crtmpserver to package h264 and aac into ts streams and play them on the iPad through HTML5, because there is no ready-to-useCodeFor reference, the packaging code is all handwritten. For the packaging format, refer to ISO/ice 18318-1.pdf. During this period, I encountered many problems and took a lot of detours. The standard-compliant ts may not be able to play on the iPad, but the
you have time to video, you can use a timer (reduce power consumption) to open a thread.
H264 and AMR-NB are transplanted from opencore, the mobile phone with baseline enough. If you want to support AAC, you can use the AAC decoder of Helix or opencore AAC decoder.
The latency caused by JNI decoding and re-transmission to the display on the Java layer is intoler
Use FAAD decoding AAC audio for PCM data flow can refer to the following articleHttp://blog.csdn.net/gavinr/article/details/6959198#reply/** * FAADDEC.C * use Faad Library to decode AAC, only can decode frame with ADTS head */#include Previously encoded with FAAC a PCM data (source data is 16000 sample rate, single channel, 16-bit sampling), the parameters set by encoding is also 16000 sample rate, single c
Author: Zhang Yu
Source: http://blog.sina.com.cn/u/465bdf0b010002t0
24-hour master of AAC File FormatToday we have a new task to save the RTSP audio stream as a file in AAC format.I have searched the internet for a long time and have not found the AAC file format.Later, I consulted a colleague who specializes in coding and decoding. There may be related informati
To use mp4v2 for MP4 video recording, prepare the following information:
1. Obtain the mp4v2 source code and compile it into a library file. For mp4v2 compilation, see the previous article.Android compiles mp4v2 2.0.0 to generate a dynamic library;
2. Obtain the SPS and PPS data in h264 data. If not, you can view the previous article.Click the open link;
3. Obtain audio decoding information, which is used when mp4settrackesconfiguration is called. The specific method is obtained through faac. fa
stream based (mpegvideo, mpegaudio) and this is the only method to use them because you cannot know the compressed data size before analysing it. BUT some other codecs (msmpeg4, mpeg4) are inherently frame based, so you must call them with all the data for one frame exactly. You must also initialize 'width' and 'height' before initializing them. */ /* NOTE2: some codecs allow the raw parameters (frame size, sample rate
Wiki:https://github.com/simple-rtmp-server/srs/wiki/v2_cn_deliveryhttpstream Note that you must use the HTTP Server that the SRS comes with One, HTTP FLV Live Stream Http_server {enabled on; Listen 8080;}Vhost Your_vhost {http_remux {enabled on;Mount [vhost]/[app]/[stream].flv; }} Second, HTTP TS Live Stream Http_server {enabled on; Listen 8080;}Vhost Your_vhost {http_remux {enabled on;Mount [Vhost]/[app]/[stream].ts; }} Third, HTTP MP3 Live StreamHttp_server {enabled on; Listen 8080;}Vhost Your
The MP4 recording program is modified according to the mpeg4ip-1.5.0.1 \ Server \ mp4live \ file_mp4_recorder.cpp file in mpeg4ip. The program supports h264 + AAC (raw stream) writing method, using the dynamic library mp4v2-2.0.0, do not use the older version of mpeg4ip, because it will be efficient when recording large files, the following describes some mp4v2 interfaces.
Mp4filehandle mp4create (const char * filename, uint32_t flags)Function: Creat
============ Problem Description ============How do I play the AAC audio that the network transmits over in Android? Each package has a ADTS head.============ Solution 1============If it is an application, set MediaPlayer Setdatasource directly to the URI of the network audio, and then Prepareasync (), prepare after the completion of the call to start is not OK?The framework layer is also to use Aacdecoder decoding playback, either with software decod
During this period of time, the work needs are converted from FLV stream to standard h264 and aac on rtmp. The server is an open-source project crtmpserver, client flex writing, video encoding h264, and audio encoding AAC, some protocol-related items are recorded as follows.
1. Analyze FLV data
Let's take a FLV file to analyze the FLV data format.
FLV standard document http://www.cnblogs.com/chef/a
A small program uses faad to decode AAC audio and can only decode the frames encapsulated by ADTs. Compiled programs can be downloaded from here: http://download.csdn.net/detail/gavinr/3782154
/*** Faaddec. C * use faad library to decode AAC, only can decode frame with ADTs head */# include
[] pbaacbuffer;}Int wavtoaac (const char * wavdata, int wavdatalen, char * aacdata ){/*** Convert WAV audio data of the length of wavdatalen into AAC and write aacdata* If AAC data is successfully returned, the value-1 is returned if an error occurs.*/If (hencoder = NULL ){Return-1;}Int32 ninputsamples = 0;Int32 readpcmbytesum = 0;Int32 writeaacbytesum = 0;Int32 naacencret = 0;Int32 J = 0;Int32 K = 0;While
This time, the work of the need, on the rtmp to do the FLV stream to the standard H264, AAC conversion, the server is open source project Crtmpserver, client flex writing, Video coding H264, Audio coding AAC, now some of the agreements related to the following things record.
I. Analysis of FLV data
Let's take an FLV file to briefly analyze the format of the FLV data
FLV standard document Download address h
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;
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