aac to m4r

. */Srs_assert (!current); /* Load the default acodec from CONFIG. */Srscodecaudio Default_acodec = SRSCODECAUDIOAAC; if (true) {/* HLS is not configured Hls_acodec, default returns "AAC" */std::string DEFAULT_ACODEC_STR = _srs_config->get_hls_ac Odec (Req->vhost); if (default_acodec_str = = "mp3") {default_acodec = SrsCodecAudioMP3; Srs_info ("hls:use default MP3 Acodec"); } else if (Default_acodec_str =

:192.168.126.191 bcast:192.168.126.255 mask:255.255.255.0[[Email protected]-BIGDATA01 Test]#ifconfig | grep ' inet addr: ' | grep-v ' 127.0.0.1 ' | sed ' s/inet addr://g '192.168.126.191 bcast:192.168.126.255 mask:255.255.255.0[[Email protected]-BIGDATA01 Test]#ifconfig | grep ' inet addr: ' | grep-v ' 127.0.0.1 ' | sed ' s/inet addr://g ' | sed ' s/bcast.*//g '192.168.126.191Misunderstandinghere is a misunderstanding, think for a long time, is the difference between regular expression and wildc

corresponding *-DEV package)!After the installation is complete, the transcoding operation is possible:(1) for ordinary avi video files, you can use the command directly:Ffmpeg-y-i "Tt.avi"-vcodec xvid-s 400x240-r 29.97-b 1500-acodec aac-ac 2-ar 48000-ab 128-vol 200-f mp4 "TT.M P4 "The parameters are explained as follows:-y overwrites the output file, that is, if the Tt.mp4 file already exists, overwrite it without prompting-I "Tt.avi" input file nam

Tag: Result lis Reference cancels space AAC func return expressionThis article describes the JavaScript regular expression catalogTo create a regular expressionRegular Expression FlagUsing regular expressions in stringsMethods of regular expressionsMetacharactersSpecial charactersGreedy modeGroup To create a regular expressionvar="I love English";var=newRegExp("love","g"); 使用构造函数创建var=/love/g; 使用字面量形式 Regular Expression Flagvar=/Love/i;g 全局匹配u u

discarded, including NO. 0 media packets, 5th media packs, and NO. 0 redundant packets. The receiving FEC is unable to recover because the number of packets received is 8 plus the number of redundant packets received is 1, the total is less than the total number of media packets (10). For the second group, only two media packets are lost and can be recovered normally. The experimental results shown in Figure 5 show that the inference is correct, No. 0 media packets, 5th media packets are lost,

thought should go to hammer, in fact, this view is wrong. Low code flow, high resolution, the use of international advanced audio and video, mainly the image of the decoding standard, a comprehensive solution actually in the middle of this part has done an online live, including the industry based on some applications, online education classes, including some entertainment contacts, are through this to achieve. Now we can provide some services, news, television, film, entertainment, music, as

from http://blog.csdn.net/aphero/article/details/78316747 1. Installing x264git clone git://git.videolan.org/x264.git ./configure--enable-shared--enable-static--enable-strip--disable-cli Make sudo make install 2. Installing FFmpeg git clone git://source.ffmpeg.org/ffmpeg.git ./configure--ENABLE-GPL--enable-version3--enable-nonfree--enable-static--enable-shared--enable-ffmpeg-- Enable-ffplay--enable-ffprobe--enable-ffserver--disable-doc--disable-htmlpages--disable-manpages-- Disable-podpages--

be compatible with the older version of the platform, need to support the baseline Profile version 3.0.2. Audio encoding compression formatHe-aac/aac-lc,stereo or MP3 (MPEG-1 Audio Layer 3), stereo.Second, talk about the video streaming technology requirements1. If the video stream is longer than 10 minutes or the video stream is over 5MB in five minutes, the HTTP Live stream technology scenario needs to b

RMMP4: Mainly used in MPEG4 EncapsulationH.264 has the highest compression ratio and is mainly used for real-time online playback at low bit rates. The rmvb compression ratio is relatively low, but the quality is much better. A complete multimedia file consists of two parts: audio and video. H264 and Xvid are video encoding formats, while MP3 and aac are audio encoding formats. Subtitle files are only included.You can package video encoding and Audio

Example: FFmpeg-y-I "1.avi"-title" test "-vcodec XviD-s 368x208-r 29.97-B 1500-acodec AAC-AC 2-ar 24000-AB 128-Vol 200-f psp-muxvb 768 "output. WMV"Explanation: The preceding commands can be entered in the doscommand line or created to run in a batch file. However, the premise is that it should be executed in the directory where FFmpeg is located (the cores subdirectory under the directory where conversion is located ).Parameters: -Y Over

Example: FFmpeg-y-I "1.avi"-title" test "-vcodec XviD-s 368x208-r 29.97-B 1500-acodec AAC-AC 2-ar 24000-AB 128-Vol 200-f psp-muxvb 768 "output. WMV" Explanation: The preceding commands can be entered in the doscommand line or created to run in a batch file. However, the premise is that it should be executed in the directory where FFmpeg is located (the cores subdirectory under the directory where conversion is located ).Parameters: -Y overwrites the o

: 1800 B = RS: 600 A = control: streamid = 1 // send audio through media stream 1 A = range: Treaty = 0-72.080000 // specifies the length of the media stream. A = Length: Treaty = 72.080000 A = rtpmap: 96 MPEG4-GENERIC/32000/2 // rtpmap information, indicating the audio for AAC its sample is 32000 A = fmtp: 96 profile-level-id = 15; mode = AAC-HBr; sizelength = 13; indexlength = 3; indexdeltalength = 3; con

[Email protected] bin]# ln-s/usr/local/mpeg4ip/bin/mp4info/usr/bin//mp4infoTar zxvf gpac-0.5.0.tar.gzTar zxvf gpac_extra_libs-0.4.5.tar.gzCD GpacCp-r. /gpac_extra_libs/* extra_lib/chmod +x Configure./configureMakeMake LibMake AppsMake installWhereis Mp4boxAdd default command[Email protected] bin]# Cd/usr/bin[Email protected] bin]# ln-s/usr/local/bin/mp4box mp4box[email protected] bin]# ll m*Mail Mp4box[email protected] bin]# ll Mp4boxlrwxrwxrwx. 1 root root 26 March 17:04 Mp4box-/usr/local/gpac/

Tags: Video solution video development technology H.264 coding instant messaging development audio and video instant messaging Development Real-time audio and video communication development is also called Instant Messaging development. In short, the development of real-time audio and video communication is based on the advanced H. 264 video coding standards, AAC audio coding standards and P2P technology, the high quality, wide adaptability, di

the video cache detector VBV. It is used as the cache between the encoder and the decoder bit stream, you can set it as small as possible or reduce latency without affecting the video quality. 3. If the latency is optimized only, a large number of key frames can be inserted between video frames so that the client can decode the video stream as soon as it receives it. However, if you need to optimize the cumulative latency during transmission, try to use as few key frames as possible, that is, I

Target to synthesize the sounds in the MP3 into the MP4.Third-party libraries in use: Mp4parser,Muxing Audio/videoThe API and the process is Straight-forward: You wrap the raw format file into a appropriate track object. h264trackimpl H 264Track = new h264trackimpl (new Span class= "Pl-smi" >filedatasourceimpl ( "Video.h264 aactrackimpl aactrack = new Aactrackimpl (new filedatasourceimpl ( aac These track object is then adde

;sample_rate, 0); Av_opt_set_sample_fmt (Swr_ctx, "OUt_sample_fmt ", sample_fmt, 0); Swr_init (swr_ctx); int64_t src_nb_samples = In_frame->nb_samples;out_frame->nb_ Samples = Av_rescale_rnd (Swr_get_delay (swr_ctx,oaudio_st->codec->sample_rate) + src_nb_samples,oaudio_st- >codec->sample_rate, Oaudio_st->codec->sample_rate, av_round_up); int ret = Av_samples_alloc (out_frame- >data, out_frame->linesize[0], Icodec->streams[audio_stream_idx]->codec->channels, out_frame- >nb_samples,oaudio_st->code

The recent project used the voice coding opus, searched the Internet, very little information, and no complete tutorial, now simply record the use of opus.First introduce opusOpusO The Pus encoder is a lossy sound coding format, by the Internet Engineering Task Force (IETF) developed for real-time sound transmission on the network, in the standard format RFC 6716. Opus format is an open format that is used without any patents or restrictions. CharacteristicsOpus was formerly known as the Celt en

When using FFmpeg or LIBAV for development, most of the time the function needs to be relatively single, such as the player only need to de-reuse and decoding module, or even only need an audio or video decoder, or need to use FFMPEG for the specified format encoding, transcoding, etc. At this time if the ffmpeg is not customized, and directly from the official Web download, it may take dozens of m of hard disk space, in order to help the product of thin, need to compile a ffmpeg to meet their n

mp3 files is ima4.Find.-name '*. mp3'-execafconvert-f caff-d aac {}\;Change the data format of a single sequence file in the current directory to aac.UsedSample. Mp3 to out. cafAfconvert-f caff-d 'ima4'-c 1 sample0000out. cafFor more command usage, enter the following command:Afconvert-hConvertAIFFFormat (Uncompressed, the file becomes larger):Afconvert-f AIFF-d I8 (I8 must be capitalized)Audio conversion software Foobar2000 playerSupported playback